• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.515-519
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• 2005

Pitch period that is a important factor in speech signal processing is used in various applications such as speech recognition, speaker identification, speech analysis and synthesis. So many pitch detection algoritms have been studied until now. AMDF which is one of pitch period detection algorithms chooses the time interval from valley point to valley point as pitch period. In selection of valley point to detect pitch period, complexity of the algoritm is increased. So in this paper we proposed the simple algorithm using modified AMDF that detects global minimum valley point as pitch period of speech signal and compared existing methods with it through simulation.

• Journal of the Korea Society of Computer and Information
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• v.28 no.12
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• pp.97-104
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• 2023

In this paper, we propose a harmful bird repellent system using directional speakers. Existing sound systems for the extermination of harmful birds have the disadvantage of reducing effectiveness due to the learning effect of birds due to problems caused by noise pollution and monotonous sounds. In this paper, directional speakers are used to minimize surrounding noise. In addition, the up-down and left-right angles of the speaker driving device were freely adjusted to maximize usability. Additionally, the problem of performance degradation due to learning effects was solved by using various scanning patterns. In the future, we plan to develop a platform capable of central control by applying remote control functions and a deep learning model that can recognize bird species.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.4 no.2
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• pp.149-154
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• 2004

In this research, we selected the speech recognition to implement the electric wheelchair system as a method to control it by only using the speech and used DTW (Dynamic Time Warping), which is speaker-dependent and has a relatively high recognition rate among the speech recognitions. However, it has to have small memory and fast process speed performance under consideration of real-time. Thus, we introduced VQ (Vector Quantization) which is widely used as a compression algorithm of speaker-independent recognition, to secure fast recognition and small memory. However, we found that the recognition rate decreased after using VQ. To improve the recognition rate, we applied ART2 (Adaptive Reason Theory 2) algorithm as a post-process algorithm to obtain about 5% recognition rate improvement. To utilize ART2, we have to apply an error range. In case that the subtraction of the first distance from the second distance for each distance obtained to apply DTW is 20 or more, the error range is applied. Likewise, ART2 was applied and we could obtain fast process and high recognition rate. Moreover, since this system is a moving object, the system should be implemented as an embedded one. Thus, we selected TMS320C32 chip, which can process significantly many calculations relatively fast, to implement the embedded system. Considering that the memory is speech, we used 128kbyte-RAM and 64kbyte ROM to save large amount of data. In case of speech input, we used 16-bit stereo audio codec, securing relatively accurate data through high resolution capacity.

• Journal of the Korean Society of Marine Environment & Safety
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• v.13 no.2 s.29
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• pp.161-167
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• 2007

The goal of this paper is to develop the speech recognition system which can control the ship's auto pilot. The feature parameters predicting the speaker's intention was extracted from the sample wheel orders written in SMCP(IMO Standard Marine Communication Phrases). And we designed the post-recognition procedure based on the parameters which could make a final decision from the list of candidate words. To evaluate the effectiveness of these parameters and the procedure, the basic experiment was conducted with total 525 wheel orders. From the experimental results, the proposed pattern recognition procedure has enhanced about 42.3% over the pre-recognition procedure.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.

• Phonetics and Speech Sciences
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• v.11 no.3
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• pp.31-38
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• 2019

The current electromagnetic articulography study analyzes several articulatory measures and examines whether, and if so, how they are interconnected, with a focus on cluster types and an additional consideration of speech rates and morphosyntactic contexts. Using articulatory data on non-assimilating contexts from three Seoul-Korean speakers, we examine how speaker-dependent gestural overlap between C1 and C2 in a low vowel context (/a/-to-/a/) and their resulting intergestural coordination are realized. Examining three C1C2 sequences (/k(#)t/, /k(#)p/, and /p(#)t/), we found that three normalized gestural overlap measures (movement onset lag, constriction onset lag, and constriction plateau lag) were correlated with one another for all speakers. Limiting the scope of analysis to C1 velar stop (/k(#)t/ and /k(#)p/), the results are recapitulated as follows. First, for two speakers (K1 and K3), i) longer normalized constriction plateau lags (i.e., less gestural overlap) were observed in the pre-/t/ context, compared to the pre-/p/ (/k(#)t/>/k(#)p/), ii) the tongue dorsum at the constriction offset of C1 in the pre-/t/ contexts was more anterior, and iii) these two variables are correlated. Second, the three speakers consistently showed greater horizontal distance between the vertical tongue dorsum and the vertical tongue tip position in /k(#)t/ sequences when it was measured at the time of constriction onset of C2 (/k(#)t/>/k(#)p/): the tongue tip completed its constriction onset by extending further forward in the pre-/t/ contexts than the uncontrolled tongue tip articulator in the pre-/p/ contexts (/k(#)t/>/k(#)p/). Finally, most speakers demonstrated less variability in the horizontal distance of the lingual-lingual sequences, which were taken as the active articulators (/k(#)t/=/k(#)p/ for K1; /k(#)t/

• Journal of Internet of Things and Convergence
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• v.8 no.3
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• pp.41-46
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• 2022

Smart home in the Internet of Things (IoT) environment aims to provide an optimal living environment for users by connecting all devices in the home. In such a smart home environment, artificial intelligence speakers are being used as a way to manage and control all devices. The existing speaker function is changing from simple music playback to the role of an interface that controls and manages all devices in the smart home space. This study dealt with the market status and usability analysis in the US and Korea, the leader in artificial intelligence speakers. The main target companies were Amazon, Google, and Apple in the US, as well as Kakao, SKT, and KT in Korea. In addition, based on the reaction results of domestic users to artificial intelligence speakers, the derivation of major problems and directions for improvement were described.

• The Journal of the Acoustical Society of Korea
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• v.28 no.5
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• pp.483-490
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• 2009

In this paper, we develop a speech interface that has acoustic echo cancelling and barge-in functionalities in the car environment. In the echo canceller, DT (Double-Talk) detection algorithm using the correlation coefficients between reference and desired signals can make DT detection errors often in the background noise. We reduce the DT detection errors by using the average power of noise and echo estimated from the input signal. In addition, to make it possible for drivers to give speech command to the system by interrupting the speaker output, barge-in functionality is implemented with the combination of DT detection and appropriate gain control of the speaker output. Through the computer simulation with the assumed car environment and experiment in the real laboratory environment, implemented speech interface has shown good performance in removing acoustic echo signals in the noisy environment with proper operation of barge-in functionality.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.61 no.7
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• pp.1069-1076
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• 2012

We use largely two methods, how to control the noise of the KTX, they are the passive noise control method and the active noise control method. The passive noise control has been used in a variety of ways since the KTX opening day, but lately it has shown the technical limitations by being dropped sharply. So, it is getting important to conduct the research about ANC that is able to reduce the ambient noise when the environmental-factor changes and be installed easily. To reduce a three-dimensional closed-space sound field like a car of a high-speed rail is hard to do using single channel ANC control system. Therefore we have to model the paths of the noise exactly for reducing the noise. And the control speakers and the error mics should be designed for optimal position. In this paper, we designed the transfer functions for modeling the noise paths under the influence of the distance between control speakers & error mics and primary noise speaker in TEST-BED where there is modeled as actual interior of KTX. We have made the modeling and the simulations of interior environment of KTX car by using three frequency bands of 120Hz, 280Hz, 360Hz. After the modeling, we compared the performance of active noise control and also we analyzed what to affect with difference in distance. After comparing of the performance using Pure Tone 120Hz, 280Hz, 360Hz at each modeling and then we simulated ANC for KTX's interior noise which we measured really and analyzed.