• Journal of Power System Engineering
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• v.9 no.3
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• pp.26-32
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• 2005

The volumetric efficiency is significantly affected by the behavior of pressure wave in induction system and exhaust pipe. By the motion of the piston, there exist pressure fluctuation in induction system which produce waves. Waves are propagated along a pipe bi-directional as they propagated through it, making compression wave and rare-faction(expansion) wave. These wave phenomena can affect to the volumetric efficiency. As a method of improvement of the volumetric efficiency, fuel economy and pollutant emission reduction particularly in low engine speeds, a side-branch additional tunable helmholtz resonator on the secondary pipe of intake system is proposed by use of their acoustic vibrations. Some of results are presented which deal with their physical phenomena for the wave action of intake system in a four-stroke three cylinders diesel engine.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.21-27
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• 1999

In this paper, we propose an improved MLP-based English pronunciation dictionary generator to apply to the variable vocabulary word recognizer. The variable vocabulary word recognizer can process any words specified in Korean word lexicon dynamically determined according to the current recognition task. To extend the ability of the system to task for English words, it is necessary to build a pronunciation dictionary generator to be able to process words not included in a predefined lexicon, such as proper nouns. In order to build the English pronunciation dictionary generator, we use context-dependent grapheme-to-phoneme multi-layer perceptron(MLP) architecture for each grapheme. To train each MLP, it is necessary to obtain grapheme-to-phoneme training data from general pronunciation dictionary. To automate the process, we use dynamic programming(DP) algorithm with some distance metrics. For training and testing the grapheme-to-phoneme MLPs, we use general English pronunciation dictionary with about 110 thousand words. With 26 MLPs each having 30 to 50 hidden nodes and the exception grapheme lexicon, we obtained the word accuracy of 72.8% for the 110 thousand words superior to rule-based method showing the word accuracy of 24.0%.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.430-437
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• 2002

Having a variety of good characteristics against other pattern recognition techniques, the multilayer perceptron (MLP) has been widely used in speech recognition and speaker recognition. But, it is known that the error backpropagation (EBP) algorithm that MLP uses in learning has the defect that requires restricts long learning time, and it restricts severely the applications like speaker recognition and speaker adaptation requiring real time processing. Because the learning data for pattern recognition contain high redundancy, in order to increase the learning speed it is very effective to use the online-based learning methods, which update the weight vector of the MLP by the pattern. A typical online EBP algorithm applies the fixed learning rate for each update of the weight vector. Though a large amount of speedup with the online EBP can be obtained by choosing the appropriate fixed rate, firing the rate leads to the problem that the algorithm cannot respond effectively to different learning phases as the phases change and the number of patterns contributing to learning decreases. To solve this problem, this paper proposes a Changing rate and Omitting patterns in Instant Learning (COIL) method to apply the variable rate and the only patterns necessary to the learning phase when the phases come to change. In this paper, experimentations are conducted for speaker verification and speech recognition, and results are presented to verify the performance of the COIL.

• The Journal of the Acoustical Society of Korea
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• v.25 no.7
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• pp.319-324
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• 2006

We Propose two new methods for non-keyword modeling to improve the performance of speaker- and vocabulary-independent keyword spotting system. The first method is decision tree clustering of monophone at the state level instead of monophone clustering method based on K-means algorithm. The second method is multi-state multiple mixture modeling at the syllable level rather than single state multiple mixture model for the non-keyword. To evaluate our method, we used the ETRI speech DB for training and keyword spotting test (closed test) . We also conduct an open test to spot 100 keywords with 400 sentences uttered by 4 speakers in an of fce environment. The experimental results showed that the decision tree-based state clustering method improve 28%/29% (closed/open test) than the monophone clustering method based K-means algorithm in keyword spotting. And multi-state non-keyword modeling at the syllable level improve 22%/2% (closed/open test) than single state model for the non-keyword. These results show that two proposed methods achieve the improvement of keyword spotting performance.

• The Journal of the Acoustical Society of Korea
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• v.30 no.1
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• pp.33-41
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• 2011

To substitute TLS in the hybrid system which is combined with Sagnac interferometer and fiber bragg grating (FBG) it is necessary to investigate how the laser source (TLS and CW) and sensor material variate the response of fiber optic sensor. Two different hollow cylinder type mandrel materials are proposed which are PTFE and PTFE+carbon and 18 m optical fiber is wounded at the mandrel surface. CW laser source experiments had been done in the oil tank which is filled with transformer oil in the 1 kHz~20 kHz frequency range. Also Sagnac interferometer fiber optic sensor is combined with FBG called hybrid system and TLS used as a light source. Based on the experimental results PTFE sensor showed more higher magnitude of detection signal rather than carbon sensor and this result is agreement with the McMahon's theoretical results. Phase variation is inversely proportional to the elastic modulus of the mandrel material. In PTFE fiber sensor, tunable laser source showed more higher performance rather than CW case. Therefore, TLS fiber optic sensor can be applied to the hybrid system which is combined with Sagnac and FBG.

• Journal of Navigation and Port Research
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• v.39 no.4
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• pp.293-297
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• 2015

Underwater acoustic communications has been limited use for military purposes in the past. However, the fields of underwater applications expend to detection, submarine and communication in recent. The excessive multipath encountered in underwater acoustic communication channel is creating inter symbol interference, which is limiting factor to achieve a high data rate and bit error rate performance. To improve the performance of a received signal in underwater communication, many researchers have been studied for channel coding scheme with excellent performance at low SNR. In this paper, we applied BCJR decoder based ( 2,1,7 ) convolution codes and to compensate for the distorted data induced by the multipath, we applying the turbo equalization method. Through the underwater experiment on the Gyeungcheun lake located in Mungyeng city, we confirmed that turbo equalization structure of BCJR has better performance than hard decision and soft decision of Viterbi decoding. We also confirmed that the error rate of decoder input is less than error rate of $10^{-1}$, all the data is decoded. We achieved sucess rate of 83% through the experiment.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.29-37
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• 1995

A resonant frequency of piezoelectric transducer depends remarkably on the electric impedance connected to the vibrator. In this paper, using this effect of frequency controllable two layered PZT ultrasonic transducer is designed and its acoustic characteristics are analyzed by a new transmission model equivalent circuit. The theoretical and the experimental results of the electric impedance effect on the resonant frequency variation were compared and both results showed a good consistency each other. The resonant frequency has been controlled continuously in the wide frequency range of 180kHz~580kHz and the effective attenuations were less than 7dB in the frequency range of 330kHz~470kHz.

• The Journal of the Acoustical Society of Korea
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• v.33 no.4
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• pp.262-267
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• 2014

Recently, long pulses are transmitted for target detection in active sonar application. Matched filtering implemented by simple convolution algorithm, requires massive computational power for long replica. The computational loads are reduced significantly by implementing the convolution in the frequency domain with overlap add method, but the performance degrades for specified input/output system delay which constrains the size of FFT function. For performance improvement, the replica could be partitioned into uniform blocks (FDL) by re-using IFFT operations, or variable blocks of increasing length (MC) by using the largest possible blocks to calculate the convolution. In this paper, by combining the strong points of the two methods, we propose a new filter partition structure that allows for further optimization of the previous two methods.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.5-10
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• 2002

This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.