• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.61-67
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• 2004

This paper describes Three-Dimensional Listening Simulation System (3D-LSS) which is to analyze whistle sound related marine casualties, and is to evaluate the accident situations using 3D sound by Head Related Transfer Function. At first, the three-dimensional listening model from the analysis of accident situations is proposed, and then the reproduction and evaluation methods of 3D sounds are also discussed. The system is designed to explain the accident situations and to simulate the possible situations with GUI based graphics and 3D sound reproduction. The evaluation experiments using 3D-LSS are carried out with six cases that did not known whether it is true or not the blast and listening of the whistle sound between two vessels. As results of psychological assessments by five subjects, the six cases can be analyzed clearly by visual images and audio sounds, thus the usability of 3D-LSS as one of the judgment assistant system of marine casualty is verified.

• Journal of Navigation and Port Research
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• v.28 no.8
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• pp.659-666
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• 2004

This paper describes Three-Dimensional Listening Simulation System (3D-LSS) which is to analyze whistle sound related marine casualties, and is to evaluate the accident situations using 3D sound by Head Related Transfer Function At first, the hree-dimensional listening model from the analysis of accident situations is proposed, and then the reproduction and evaluation methods of 3D sounds are also discussed. The system is designed to explain the accident situations and to simulate the possible situations with GUI based graphics and 3D sound reproduction. The evaluation experiments using 3D-LSS are carried out with six cases that did not known whether it is true or not the blast and listening of the whistle sound between two vessels. As results of psychological assessments by five subjects, the six cases can be analyzed clearly by visual images and audio sounds, thus the usability of 3D-LSS as one of the judgment assistant system of marine casualty is verified.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1740-1748
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• 2014

This paper focuses on listening test and acoustic analysis of patients' speech with velopharyngeal insufficiency (VPI) and normal speakers' simulation speech. In this research, a set consisting of 50-words, vowels and single syllables is determined for speech database construction. A web-based listening evaluation system is developed for a convenient/automated evaluation procedure. The analysis results show the trend of incorrect recognition for VPI speech and the one for simulation speech are similar. Such similarity is also confirmed by comparing the formant locations of vowel and spectrum of consonant sounds. These results show that the simulation method for VPI speech is effective at generating the speech signals similar to actual VPI patient's speech. It is expected that the simulation speech data can be effectively employed for our future work such as acoustic model adaptation.

• Journal of Broadcast Engineering
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• v.21 no.5
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• pp.792-802
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• 2016

Real time verification is necessary, since there are several features that cannot be verified through design simulation in the design of multiband soundbar system. And then this paper describes an implementation of an FPGA-based real-time verification system for a soundbar SoC with an embedded processor. It is verified a real-time performance test and a listening test which are several features in the design stage that cannot be verified through a design simulation. The measurement of quantitative specifications such as SNR, THD+N, frequency response, etc. as well as the listening test were performed through the implemented FPGA system, and it was verified that test results satisfied the target specifications.

• Journal of information and communication convergence engineering
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• v.9 no.2
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• pp.161-165
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• 2011

Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. Separating technique for singing voice from music accompaniment is very useful in such equipment. We propose a system to separate singing voice from music accompaniment for stereo recordings. Our system consists of three stages. The first stage is a spectral change detector. The second stage classifies an input into vocal and non vocal portions by using GMM classifier. The last stage is a selective frequency separation stage. The results of removed by listening test from the results for computer based extraction simulation, spectrogram results show separation task successfully. Listening test with extracted MR from proposed system show vocal separating and removal task successfully.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.3
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• pp.1393-1405
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• 2017

It is well known that duty-cycling control by dynamically adjusting the polling interval according to the traffic loads can effectively achieve power saving in wireless sensor networks. Thus, there has been a significant research effort in developing polling interval adaptation schemes. Especially, Dynamic Low Power Listening (DLPL) scheme is one of the most widely adopted open-looping polling interval adaptation techniques in wireless sensor networks. In DLPL scheme, if consecutive idle (busy) samplings reach a given fixed threshold, the polling interval is increased (decreased). However, due to the trial-and-error based approach, it may significantly deteriorate the system performance depending on given threshold parameters. In this paper, we propose a novel DLPL scheme, called SDL (Sequential hypothesis testing based Dynamic LPL), which employs sequential hypothesis testing to decide whether to change the polling interval conforming to various traffic conditions. Simulation results show that SDL achieves substantial power saving over state-of-the-art DLPL schemes.

• ETRI Journal
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• v.38 no.3
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• pp.425-434
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• 2016

A novel average magnitude difference function (AMDF)-based pitch detection scheme (PDS) is proposed to achieve better performance in speech quality. A performance evaluation of the proposed PDS is carried out through both a simulation and a real-time implementation of a speech analysis-synthesis system. The parameters used to compare the performance of the proposed PDS with that of PDSs that are based on either a cepstrum, an autocorrelation function (ACF), an AMDF, or circular AMDF (CAMDF) methods are as follows: percentage gross pitch error (%GPE); a subjective listening test; an objective speech quality assessment; a speech intelligibility test; a synthesized speech waveform; computation time; and memory consumption. The proposed PDS results in lower %GPE and better synthesized speech quality and intelligibility for different speech signals as compared to the cepstrum-, ACF-, AMDF-, and CAMDF-based PDSs. The computational time of the proposed PDS is also less than that for the cepstrum-, ACF-, and CAMDF-based PDSs. Moreover, the total memory consumed by the proposed PDS is less than that for the ACF- and cepstrum-based PDSs.

• Journal of Communications and Networks
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• v.14 no.2
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• pp.179-187
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• 2012

Power saving is one of the most important features that extends the lifetime of portable devices in mobile wireless networks. The IEEE 802.16e mobile broadband wireless access system adopts a power saving mechanism with a binary truncated exponent algorithm for determining sleep intervals. When using this standard power saving scheme, there is often a delay before data packets are received at the mobile subscriber station (MSS). In order to extend the lifetime of a MSS, the battery energy must be used efficiently. This paper presents a dynamically alternating sleep interval scheduling algorithm as a solution to deal with the power consumption problem. We take into account different traffic classes and schedule a proper sequence of power saving classes. The window size of the sleep interval is calculated dynamically according to the packet arrival rate. We make a tradeoff between the power consumption and packet delay. The method achieves the goal of efficiently reducing the listening window size, which leads to increased power saving. The performance of our proposed scheme is compared to that of the standard power saving scheme. Simulation results demonstrate the superior performance of our power saving scheme and its ability to strike the appropriate performance balance between power saving and packet delay for a MSS in an IEEE 802.16e mobile broadband wireless access system.

• Journal of the Korean Institute of Telematics and Electronics
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• v.22 no.5
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• pp.29-38
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• 1985

In this paper we have designed and implemented in real time a simple, efficient and flexible AOPCM cosec using a high speed digital processor, NEC 7720. For ADPCM system, we have used an instantaneous adaptive quantizer and a first-order fixed predictor. The software for NEC 7720 has been developed and it was found that the NEC 7720 was capable of performing the entire ADPCAt algorithm for 4 channels in real time as optimizing the program. Computer simulation has born made to investigate a computational accuracr of NEC 7720 and to de-termine necessary parameters for a ADPCM codec. Real telephone speech, RC-shaped Gaussian noise and 1004 Hz tone signal were used for simulation. In simulation, the parameters werc optimized from the computed SNR and the informal listening test. The developed software was tested in real time operation using a hardware emulator for NEC 7720. It took a maximum 23.25$\mu$s to encode one sample and 113.5$\mu$s, including all the necessary 1/0 operations, to encode 4 channels. In the case of decoding process, it took 24.75$\mu$s to decode one sample and 119.5$\mu$s to decode 4 channels.

• The KIPS Transactions:PartC
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• v.17C no.6
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• pp.471-484
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• 2010

In the wireless communication systems, the power consumption of a mobile device is very important issue due to its battery limitations. Hence most of the standards for wireless networks including a mobile WiMAX system are supporting their own power saving mode in way that a mobile device is able to reduce its energy usage while in the mode. However, those standards just define the arrangement of special time intervals, called a paging listening interval, during which the device needs to receive the paging-related control messages, and they do not specify how to effectively reduce the power in many different network environments. This means the amount of power spent by the device is very dependent on the implementations of individual device-vendors, and undesirable paging loss may happen according to the channel conditions. To reduce unnecessary power usage and the risk of paging loss, this paper proposes the effective frequency/BS selection algorithm applicable to a mobile device operating in the power saving mode, which serves the device with better BS based on the tracking for channel variation. This algorithm consists of the channel estimation phase during each paging listening interval, the tracking phase for the measured results, the frequency reselection phase based on the tracking activity, and the preferred BS reselection phase. Thus the proposed method can improve the paging performance while the device is moving in the network. Also the simulation result shows that the presented scheme is superior to other candidates in energy efficiency due to the channel-adaptive frequency/BS selection.