• 대한전기학회:학술대회논문집
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• 대한전기학회 2006년도 심포지엄 논문집 정보 및 제어부문
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• pp.261-263
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• 2006

Nowadays, technologies such as RFID, sensor network makes our life comfortable more and more. In this paper we propose a wearable computing system for blind and deaf person who can be easily out of sight from our technology. We are making a wearable computing system that is consisted of embedded board to processing data, ultrasonic sensors to get distance data and motors that make vibration as a signal to see the screen for a deaf person. This system offers environmental informations by text and voice. For example, distance data from a obstacle to a person are calculated by data compounding module using sensed ultrasonic reflection time. This data is converted to text or voice by main processing module, and are serviced to a handicapped person. Furthermore we will extend this system using a voice recognition module and text to voice convertor module to help communication among the blind and deaf persons.

• 한국항행학회논문지
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• 제21권1호
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• pp.113-118
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• 2017

최근 모바일기기 및 네트워크에 대한 관심이 높아짐에 따라 기존의 IP (internet protocol) 망을 이용하여 음성데이터를 전송하는 기술인 VoIP (voice over internet protocol)서비스가 급속히 확산됨에 따라 무선 인터넷망을 활용하여 언제 어디서나 저렴한 음성 통화 서비스가 가능해졌다. 그리고 디지털방송서비스가 보급되면서 방송과 통신의 융합을 통해 광대역케이블망을 이용하는 HFC (hybrid fiber coaxial)망 기술은 기존의 통신시스템 및 망설비를 활용하여 양방향 방송서비스 및 인터넷, 전화 등 다양한 신규 서비스를 제공하고 있다. 따라서 실제 HFC 인터넷서비스망에서 음성데이터의 품질보장을 위해 VoCM에 UGS-AD를 MTA에는 RTPS를 적용하면 실제 상용 HFC 인터넷서비스망에서 문제가 되는 협소한 상향대역에서의 음성데이터 전송을 원활히 수행할 수 있음을 확인하였으며, HFC 인터넷서비스 망에서 음성데이터의 QoS개선을 통해 기존 대비 개선된 VoIP서비스를 구현하는 방안을 제시하였다.

• International journal of advanced smart convergence
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• 제11권1호
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• pp.55-63
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• 2022

Although research on the leader's or speaker's voice has been continuously conducted, existing research has a single point of view. Sound analysis of voice characteristics has been studied from engineering perspectives, and leadership trait theory has been studied from a business perspective. Convergence studies on leader voice and member cognition are being attempted today. Convergence research on voice has a positive effect on refinement of voice analysis, diversification of voice use, and establishment of voice utilization strategy. This study explains the current flow of research on convergence between speaker's voice and listener's perception, and suggests a direction for the future development of voice fusion research. Furthermore, in connection with AI in the 4th industrial age, new attempts for voice research are sought. First, advances in AI focus on strategically generating the voices needed for individual situations. Second, the voice corrected in real time will support the leader and speaker to utilize the desired voice type. Third, voices through AI based on big data will affect the cognition, attitude and behavior of individual listeners who members, customers, and students in more diverse situations. The purpose and significance of this study is to suggest the way to research the leader's voice recognized by members, and to suggest a method that can be applied in various situations.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2007년도 심포지엄 논문집 정보 및 제어부문
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• pp.275-277
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• 2007

Recent activation of DMB has enabled various high quality video, audio and data services. And there are various user facilities functions using digital data transmission. One of the various user facilities functions is EPG(Electronic Program Guide). EPG supports schedule of programs on screen for. audiences. EPG is composed to time, title, channel, genre etc. Users can select a program what they want to browsing. Currently EPG services are displaying program schedule on screen visually and make users to input ke:ywords with keypads, remote control devices or touch screen etc. However, this approach could cause a serious restriction to some users like to drivers or visually handicapped persons. A standard for a voice EPG to T-DMB is proposed. This method must be transferred VoiceXML based EPG files from the transmitter to receivers. This approach has a problem to process a standardization because the transmitter and receivers should be modified. We proposed and implemented a voice EPG platform that generates the voice EPG files from T-DMB SI without transferring voice EPG file from the transmitter.

• 제어로봇시스템학회논문지
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• 제22권10호
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• pp.797-802
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• 2016

This paper proposes a noble voice recognition method based on an adaptive MFCC and deep learning for embedded systems. To enhance the recognition ratio of the proposed voice recognizer, ambient noise mixed into the voice signal has to be eliminated. However, noise filtering processes, which may damage voice data, diminishes the recognition ratio. In this paper, a filter has been designed for the frequency range within a voice signal, and imposed weights are used to reduce data deterioration. In addition, a deep learning algorithm, which does not require a database in the recognition algorithm, has been adapted for embedded systems, which inherently require small amounts of memory. The experimental results suggest that the proposed deep learning algorithm and HMM voice recognizer, utilizing the proposed adaptive MFCC algorithm, perform better than conventional MFCC algorithms in its recognition ratio within a noisy environment.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(1)
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• pp.189-192
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• 2000

This paper proposes a fuzzy-based random access controller with a superimposed frame structure (F$^2$RAC) fur voice/data-integrated wireless networks. F$^2$RAC adopts mini-slot technique for reducing contention cost, and these mini-slots of which number may dynamically vary from one frame to the next as a function of the traffic load are further partitioned into two regions for access requests coming from voice and data traffic with their respective QoS requirements. And F$^2$RAC is designed to properly determine the access regions and permission probabilities for enhancing the data packet delay while ensuring the voice packet dropping probability constraint. It mainly consists of the estimator with Pseudo-Bayesian algorithm and fuzzy logic controller with Sugeno-type of fuzzy rules. Simulation results prove that F$^2$RAC can guarantee QoS requirement of voice and provide the highest throughput efficiency and the smallest data packet delay amongst the different alternatives including PRMA［1］, IPRMA［2］, and SIR［3］.

• 정보처리학회논문지D
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• 제15D권2호
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• pp.257-262
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• 2008

본 논문에서는 음성통화를 위해 CDMA 네트워크와 데이터 통신을 위해 와이브로 네트워크를 동시에 사용하는 임베디드 리눅스 기반의 듀얼모드 응용 서비스인 개인 오디오 레코더의 구현에 대해서 기술한다. 개인 오디오 레코더는 듀얼모드 지원 단말에 탑재된 클라이언트에서 음성 녹음을 시작하면, 송신자와 수신자의 CDMA 음성 데이터가 와이브로 네트워크를 통해 인터넷 상의 저장 서버로 전달된다. 개인 오디오 레코더 서버는 통화 번호 및 통화 시간을 기준으로 음성 데이터를 서버에 저장하게 된다. 구현된 개인 오디오 레코더는 단말의 저장공간이 부족한 환경에서도 음성 통화 내용을 저장할 수 있도록 한다. 그리고, 개인 오디오 레코더는 서버에 저장된 통화 목록을 검색하여, 특정 통화 내용을 재생할 수 있다.

• 산업경영시스템학회지
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• 제29권4호
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• pp.100-106
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• 2006

We analyze a voice/data integrated traffic model of the cellular mobile communication system with hybrid guard channels for voice and handoff calls. In a multi-service integrated wireless environment, quality of service guarantee is crucial for smooth transportation of real time information. Real time voice traffic requires a guaranteed upper bounded on both delay and packet error rate, whereas data traffic does not. Voice traffic has high transmission priority over data packets. Thus one of the important problems is the design of admission control schemes which can efficiently accommodate the differential quality of service requirements. In this paper, a hybrid guard channel scheme is considered in which arriving calls are assigned channels as long as the number of busy channels in the cell is below a predetermined first threshold. When the number of busy channels reaches the first threshold, new originating data calls are queued in the infinite data buffer. Then reaches second threshold, only handoff calls are assigned the remaining channels and new originating voice calls are blocked. We evaluate the system by a two-dimensional Markov chain approach and generating function method and obtain performance measures included blocking probability and forced termination probability.

• 음성과학
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• 제10권4호
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• pp.201-219
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• 2003

Background: The perceptual assessment is generally performed by the voice specialist. The objective evaluation is performed in a voice laboratory. Research in voice laboratories has generated a variety of different objective tests and parameters. The perceptual evaluation is one of the most controversial topics in voice research. Review of literature reveals a wide variety of rating scales and reliability data fluctuating from study to study. Unfortunately, there is no widely accepted valid method for classifying voice disorders and assessing outcome after voice treatment. Objectives: The goals of this research were to identify important objective acoustic parameters of vocal quality, and to establish an objective and quantitative correlate of the perceived vocal quality. Materials and Methods : We evaluated the voice analyzed data from 122 dysphonic patients and 20 normal volunteers. A computerized speech lab. 4300B(CSL) was used to carry out the analysis of each voice sample. Results: Three dysphonia severity indices(DSI) were created using discriminant analysis. DSI is based on the weighted combination of the following selected set of acoustic parameters: absolute jitter(Jita in us), smoothed pitch period perturbation (sPPQ in %), amplitude perturbation quotient(APQ in %), soft phonation index(SPI), average fundamental frequency(Fo in Hz), lowest fundamental frequency(Flo in Hz), and smoothed amplitude perturbation quotient(sAPQ in %). The DSI, being the discriminating rule calculated by the logistic regression, consists of three equation based on statistically significant acoustic parameters. Three DSI were created to reflects best the degree of hoarseness as expressed by G from the GRBAS scale. The more positive this DSI is for a patient, the worse the vocal quality. The more it is negative, the better it is. The effect of sex is included implicitly in the DSI-1 and DSI-2, so that a separate DSI-1 and DSI-2 for males and females need not be used. The DSI is objective because no perceptual input is required for its calculation. Conculsion : This research demonstrates that the voice function values calculated from three different multivariate objective dysphonia severity indices are significantly associated with subjective voice assessments. These multivariate objective dysphonia severity indices may be appropriate for use in clinical trials and outcomes research on treatment effectiveness for voice disorders.

• 대한전자공학회논문지TC
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• 제40권9호
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• pp.373-383
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• 2003

본 논문에서는 다양한 서비스가 복합적으로 운용되고 있는 인터넷 망에서PCM 및 ADPCM으로 압축된 음성 데이터를 전송할 경우, 패킷 크기와 한계 지연시간의 변화가 수신측의 음질에 미치는 영향을 분석하고 주어진 한계 지연시간에 대하여 최고의 음질을 제공하기 위한 전송 패킷의 크기에 대하여 논한다. 결과를 얻기 위한 실험은 공중 인터넷 망을 통하여 연결된 두 대의 PC 상에서 이루어졌다. 송신측은 마이크로부터의 음성신호를 PCM 및 ADPCM으로 부호화하고 이를 UDP 패킷을 이용하여 전송하였으며, 수신 측에서는 망에서 발생하는 전송 지연 및 패킷 손실 등을 거친 패킷이 음성신호로 재생된다. 음질 평가를 위하여 송수신 음성 데이터를 수집하여 오프라인에서 비교하며, 알고리즘으로는 객관성을 유지하기 위하여 MNB 방법을 이용하였다. 실험 결과에 의하면, 40Kbps, 32Kbps, 16Kbps의 ADPCM으로 압축된 음성의 전송에서 한계 전송 지연을 100ms로 하였을 경우 음질 열화를 최소화하기 위하여는 패킷 크기의 하한이 각각 300, 400, 500바이트, 패킷 크기의 상한은 공히 1200바이트인 것이 요구된다.