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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 29, Issue 8 - Nov 2010
Volume 29, Issue 7 - Oct 2010
Volume 29, Issue 6 - Aug 2010
Volume 29, Issue 5 - Jul 2010
Volume 29, Issue 4 - May 2010
Volume 29, Issue 3 - Apr 2010
Volume 29, Issue 2 - Feb 2010
Volume 29, Issue 1 - Jan 2010
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Calculation of the Mutual Radiation Impedance by the Spatial Convolution in the Cylindrical Structure
Bok, Tae-Hoon ; Li, Ying ; Paeng, Dong-Guk ; Lee, Jong-Kil ; Shin, Ku-Kyun ; Joh, Chee-Yong ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 1~9
The mutual radiation impedance was calculated using the spatial convolution in the cylindrical structure. The Cartesian coordinate was transformed into the cylindrical coordinate using the spatial convolution for the cylindrical array structure. This method cannot consider the cylindrical baffle, but can reduce the computation time. The error for not considering the cylindrical baffle was analyzed by the comparison of the spatial convolution method with the quadruple integration method in the cylindrical structure. The mutual radiation resistance in the cylindrical structure was compared with the one in the planar baffle. Based on two kinds of the comparison, we presented the error of the suggesting method in this paper, confirming that the spatial convolution method could be applied to compute the mutual radiation impedance in the cylindrical structure at certain conditions.
Performance Analysis of Own Ship Noise Cancellation in Hull Mounted Sonar System Using Adaptive Filter
Yoon, Kyung-Sik ; Jung, Tae-Jin ; Lee, Kyun-Kyung ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 10~17
In a passive sonar, the improvement of detection performance by using noise cancellation is usually a important problem. In this paper, we have analyzed the own-ship noise cancellation in the two operation modes which are used in the HMS system. In the operator mode, an adaptive line enhancer(ALE) is applied to improve the tonal detection by using broadband noise cancellation and the normalized least mean square(NLMS) algorithm is applied to the design of an adaptive filter. The reference input that is correlated with a primary input can be used to remove the noise incident on the observation directionin the automatic mode. Computer simulations with real sea that data show that the proposed adaptive noise canceller has good performance in passive detection under HMS operation.
Low-Complexity VFF-RLS Algorithm Using Normalization Technique
Lee, Seok-Jin ; Lim, Jun-Seok ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 18~23
The RLS (Recursive Least Squares) method is a broadly used adaptive algorithm for signal processing in electronic engineering. The RLS algorithm shows a good performance and a fast adaptation within a stationary environment, but it shows a Poor performance within a non-stationary environment because the method has a fixed forgetting factor. In order to enhance 'tracking' performances, BLS methods with an adaptive forgetting factor had been developed. This method shows a good tracking performance, however, it suffers from heavy computational loads. Therefore, we propose a modified AFF-RLS which has relatively low complexity m this paper.
High Directivity Sound Beamforming Algorithm
Kim, Seona-Woo ; Hur, Yoo-Mi ; Park, Young-Chul ; Youn, Dae-Hee ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 24~33
This paper proposes a technique of sound beamforming that can generate high-directive sound beams, and this paper also presents applications of the proposed algorithm to multi-channel 3D sound systems. The proposed algorithm consists of two phases: first, optimum weights maximizing a sound pressure level ratio between the target and control acoustic regions are designed, and later, the directivity of the pre-designed sound beam is iteratively enhanced by modifying the covariance matrix. The proposed method was evaluated under various situations, and the results showed that it could provide more focused sound beams than the conventional methods.
The Performance Improvement of G.729 PLC in Situation of Consecutive Frame Loss
Hong, Seong-Hoon ; Kim, Jin-Woo ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 34~40
As internet spread widely, various service which use the internet have been provided. One of the service is a internet phone. Its usage is increasing by the advantage of cost. But it has a falling off in quality of speech. because it use packet switching method while existing telephone use circuit switching method. Although vocoder use PLC (Packet Loss Concealment) algorithm, it has a weakness of continuous packet loss. In this paper, we propose methods to improve a lowering in quality of speech under continuous loss of packet by using PLC algorithm used in advanced G.729 and G.711. The proposed methods are LP (Linear Prediction) parameter interpolation, excitation signal reconstruction and excitation signal gain reconstruction. As a result, the proposed method shows superior performance about 11%.
On a Processing Time Reduction of Cepstrum-Based Pitch Alteration in Time-Frequency Hybrid Domain
Jo, Wang-Rae ; Kim, Jong-Kuk ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 41~47
The pitch alteration technique for voice conversion is classified in time domain, frequency domain and hybrid domain. The Hybrid domain method has a merit of clearness and natural-ness of pitch altered speech but has the major drawback of long processing time. In this paper, we proposed a new method that can reduce the processing time of pitch alteration in time-frequency hybrid domain. We omitted the bit-reversing process of FFT and IFFT in changing the processing domain. Therefore we can reduce the processing time by 86.26% to the conventional method with same quality.
The Performance Improvement of PLC by Using RTP Extension Header Data for Consecutive Frame Loss Condition in CELP Type Vocoder
Hong, Seong-Hoon ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 48~55
It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.
A Study on Improved MDL Technique for Optimization of Acoustic Model
Cho, Hoon-Young ; Kim, Sang-Hun ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 56~61
This paper describes optimization methods of acoustic models in HMM-based continuous speech recognition. Most of the conventional speech recognition systems use the same number of Gaussian mixture components for each HMM state. However, since the number of data samples available for each state is different from each other, it is possible to reduce the overall number of model parameters and the computational cost at the decoding step by optimizing the number of Gaussian mixture components. In this study, we introduced the Gaussian mixture weight term at the merging stage of Gaussian components in the minimum description length (MDL) based acoustic modeling optimization. Experimental results showed that the proposed method can obtain better ASR accuracy than the previous optimization method which does not consider the Gaussian mixture weight term.
Enhanced Adjustment Strategy of Masking Threshold for Speech Signals in Low Bit-Rate Audio Coding
Lee, Chang-Heon ; Kang, Hong-Goo ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 62~68
This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.
A Study on the Pitch Extraction Improvement Using LSP for the Synthesis of High Speech Quality
Seo, Ji-Ho ; Kim, Jong-Kuk ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 69~75
In this paper, the pitch is detected after the elimination of formant ingredients by flattening the spectrum in frequency domain. In order to remove impact of formant and transition frequency in the signal spectrum, formant envelop is made by linear interpolation with any points each sub-band and the spectrum of speech signal is compensated by the reverse of the envelop interpolated linearly after we divide frequency band into several segment based on LSP and detect the points. The experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum, Lifter methods. The method reduced the gross error rate 1.30% than the LPC method which appeared a good performance except the proposed method. Also, the proposed method showed low error rate in noise environment.
Statistical Model-Based Voice Activity Detection Using the Second-Order Conditional Maximum a Posteriori Criterion with Adapted Threshold
Kim, Sang-Kyun ; Chang, Joon-Hyuk ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 76~81
In this paper, we propose a novel approach to improve the performance of a statistical model-based voice activity detection (VAD) which is based on the second-order conditional maximum a posteriori (CMAP). In our approach, the VAD decision rule is expressed as the geometric mean of likelihood ratios (LRs) based on adapted threshold according to the speech presence probability conditioned on both the current observation and the speech activity decisions in the pervious two frames. Experimental results show that the proposed approach yields better results compared to the statistical model-based and the CMAP-based VAD using the LR test.
Spectrum Based Excitation Extraction for HMM Based Speech Synthesis System
Lee, Bong-Jin ; Kim, Seong-Woo ; Baek, Soon-Ho ; Kim, Jong-Jin ; Kang, Hong-Goo ;
The Journal of the Acoustical Society of Korea, volume 29, issue 1, 2010, Pages 82~90
This paper proposes an efficient method to enhance the quality of synthesized speech in HMM based speech synthesis system. The proposed method trains spectral parameters and excitation signals using Gaussian mixture model, and estimates appropriate excitation signals from spectral parameters during the synthesis stage. Both WB-PESQ and MUSHRA results show that the proposed method provides better speech quality than conventional HMM based speech synthesis system.