• Title/Summary/Keyword: Voice problem

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The Voice Template based User Authentication Scheme Suitable for Mobile Commerce Platform (모바일 상거래 플랫폼에 적합한 음성 템플릿 기반의 사용자 인증 기법)

  • Yun, Sung-Hyun;Koh, Hoon
    • Journal of Digital Convergence
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    • v.10 no.5
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    • pp.215-222
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    • 2012
  • A smart phone has functions of both telephone and computer. The wide spread use of smart phones has sharply increased the demand for mobile commerce. The smart phone based mobile services are available anytime, anywhere. In commercial transactions, a digital signature scheme is used to make legally binding signature to prove both integrity of commercial document and verification of the signer. Smart phones are more risky compared with personal computers on the problems of how to protect privacy information. It's also easy to let proxy user to authenticate instead of the smart phone owner. In existing password or token based schemes, the ID is not physically bound to the owner. Thus, those schemes can not solve the problem of proxy authentication. To utilize the smart phone as the platform of mobile commerce, a study on the new type of authentication scheme is needed where the scheme should provide protocol to get legally binding signature and not to authenticate proxy user. In this paper, we create the mobile ID by using both the USIM and voice template of the smart phone owner. We also design and implement the user authentication scheme based on the mobile ID.

Time Slot Exchange Protocol in a Reservation Based MAC for MANET

  • Koirala, Mamata;Ji, Qi;Choi, Jae-Ho
    • Journal of the Institute of Convergence Signal Processing
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    • v.10 no.3
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    • pp.181-185
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    • 2009
  • Recently, much attention to a self-organizing mobile ad-hoc network is escalating along with progressive deployment of wireless networks in our everyday life. Being readily deployable, the MANET (mobile ad hoc network) can find its applications to emergency medical service, customized calling service, group-based communications, and military purposes. In this paper we investigate a time slot exchange problem found in the time slot based MAC, that is designed for IEEE 802.11b interfaces composing a MANET. The paper provides a method to maintain the quality of voice call by providing a new time slot when the channel assigned for that time slot gets noisy with interferences induced from other nodes, which belong to the same and/or other subgroups. In order to assess the performance of the proposed algorithm, a set of simulations using the OPNET modeler has been performed assuming that the IEEE 802.11b interfaces are operating under a modified MAC, which is a time slot based reservation MAC implemented in the PCF part of the superframe. In a real-time voice call service over a MANET of a size 500 ${\times}$ 500 meter squares with the number of nodes up to 100, the simulation results are collected and analyzed with respect to the packet loss rate and packet delay. The results show us that the proposed time slot exchange protocol improves the quality of voice call over that of plain DCF.

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Frequency Domain Acoustic Echo Suppression Based on Boundary Condition (주파수 영역에서 구간조건을 이용한 음향학적 반향 제거)

  • Lee, Kyu-Ho;Chang, Joon-Hyuk
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.46 no.5
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    • pp.162-166
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    • 2009
  • In this paper, we propose a novel approach of an acoustic echo cancellation (AEC) algorithm which is differently adopted in the relevant period condition by the suppression parameter of a parametric wiener filter (PWF). The PWF uses the suppression parameter to compensate uncertainty of acoustic echo signal estimation. The existing PWF method using the fixed suppression parameter derives the distortion of the near-end signal at the double-talk. To solve this problem, the boundary condition is devised using decision of the double-talk detection (DTD) algorithm and voice activity detector (VAD). The boundary condition makes it possible to treat differently depending on the case of the single-talk and double-talk. According to the experimental results, the proposed approach is found to be effective for acoustic echo cancellation using the boundary condition.

Variation Measurement and Analysis of Jitter and Shimmer Parameter Value by Hemodialysis in Diabetic and Hypertensive (당뇨 및 고혈압 환자에서 혈액투석에 따른 Jitter와 Shimmer 요소값 변화 측정 및 분석)

  • Kim, Bong-Hyun;Cho, Dong-Uk
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.7B
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    • pp.834-840
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    • 2011
  • Chronic diseases is being increased attention to threatening the elements healthy life of elderly population in modem society. Especially, Chronic diseases caused by diabetes and hypertension is destroyed kidney. In this case, subjective symptom is not little. So if health is worsened, hemodialysis, artificial organs, transplant an organ etc. should be treated as a dangerous diseases. Therefor, a patients receiving hemodialysis of diabetes and hypertension studied on the effects to regularity of amplitude and rate vibration of vocal cords in hemodialysis in this paper. To do this, a diabetic and hypertensive patients don't have a problem with pronunciation selected as of the subjects and their voices collected before and after hemodialysis. We studied on the effects of voice analysis to apply regularity of amplitude and rate vibration of vocal cords. In conclusion, we extracted a result that voice after than before hemodialysis is relatively low in voice measures values a regularity of amplitude and rate vibration of vocal cords.

Investigation of Timbre-related Music Feature Learning using Separated Vocal Signals (분리된 보컬을 활용한 음색기반 음악 특성 탐색 연구)

  • Lee, Seungjin
    • Journal of Broadcast Engineering
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    • v.24 no.6
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    • pp.1024-1034
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    • 2019
  • Preference for music is determined by a variety of factors, and identifying characteristics that reflect specific factors is important for music recommendations. In this paper, we propose a method to extract the singing voice related music features reflecting various musical characteristics by using a model learned for singer identification. The model can be trained using a music source containing a background accompaniment, but it may provide degraded singer identification performance. In order to mitigate this problem, this study performs a preliminary work to separate the background accompaniment, and creates a data set composed of separated vocals by using the proven model structure that appeared in SiSEC, Signal Separation and Evaluation Campaign. Finally, we use the separated vocals to discover the singing voice related music features that reflect the singer's voice. We compare the effects of source separation against existing methods that use music source without source separation.

Noise Elimination Using Improved MFCC and Gaussian Noise Deviation Estimation

  • Sang-Yeob, Oh
    • Journal of the Korea Society of Computer and Information
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    • v.28 no.1
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    • pp.87-92
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    • 2023
  • With the continuous development of the speech recognition system, the recognition rate for speech has developed rapidly, but it has a disadvantage in that it cannot accurately recognize the voice due to the noise generated by mixing various voices with the noise in the use environment. In order to increase the vocabulary recognition rate when processing speech with environmental noise, noise must be removed. Even in the existing HMM, CHMM, GMM, and DNN applied with AI models, unexpected noise occurs or quantization noise is basically added to the digital signal. When this happens, the source signal is altered or corrupted, which lowers the recognition rate. To solve this problem, each voice In order to efficiently extract the features of the speech signal for the frame, the MFCC was improved and processed. To remove the noise from the speech signal, the noise removal method using the Gaussian model applied noise deviation estimation was improved and applied. The performance evaluation of the proposed model was processed using a cross-correlation coefficient to evaluate the accuracy of speech. As a result of evaluating the recognition rate of the proposed method, it was confirmed that the difference in the average value of the correlation coefficient was improved by 0.53 dB.

Design of Smart Glasses Platform walking guide for the visually impaired (시각장애인을 위한 보행 안내 스마트 안경 플랫폼 설계)

  • Lee, Jaebeom;Jang, Jongwook;Jang, Sungjin
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2021.10a
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    • pp.320-322
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    • 2021
  • As the world's elderly population increases, the proportion of visually impaired is also increasing, and there are still many restrictions on the use of outside activities, such as safety problems and lack of guidance information. To solve this problem, research on smart devices such as smart glasses with optical character recognition (OCR) function is being actively conducted. In this paper, we propose a system that recognizes obstacles ahead and informs information by voice, and also guides the way to the destination. Using the deep learning object recognition model Yolo, it let them to recognize the risk factors as obstacles such as stairs and Larva cones. and it also deliver the information with a voice. so you can expect that the visually impaired can do a lot of different activity even more now that system takes the visually impaired to the destination by using the directions API, voice recognition, TTS library.

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An Efficient Dynamic Bandwidth Allocation Algorithm for VoDSL Services (VoDSL 서비스를 위한 효율적인 동적 대역폭 할당 알고리즘)

  • Kim, Hoon;Park, Jong-Dae;Nam, Sang-Sig;Park, Kwang-Chae
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.1C
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    • pp.48-58
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    • 2002
  • As internet traffic increases, the problem that it should be efficiently accepted in to the existing voice network in the pending problem importantly to the existing communication corporations. The feature of next generation exchange network is made up of the form of integration network that connect data traffic(internet service. Etc) with the existing voice network and it can be showed with very diverse aspects according to the constitution time of network or the characteristics of business. The progress strategy that develop the existing circuit based communication network into packet-based communication network can be divided into two in a large scale according to the application position These are VoDSL application method(Technology that packetize access network) and softswitch technology application method(after packetizing relay network, packetizing that into the access network). In this paper, we will deduce the desirable technology that can construct packet-based next generation exchange networks in the structure of the existing communication network environment. We will perform the research on a device to offer the necessary core technique VoSDL service with realistic resolutions primarily.

A NAT Proxy Server for an Internet Telephony Service (인터넷 전화 서비스를 위한 NAT 프럭시 서버)

  • 손주영
    • Journal of KIISE:Computing Practices and Letters
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    • v.9 no.1
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    • pp.47-59
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    • 2003
  • The Internet telephony service is one of the commercially successful Internet application services. VoIP technology makes the service come true. VoIP deploys H.323 or SIP as the standard protocol for the distributed multimedia services over the Internet in which QoS is not guaranteed. VoIP carries the packetized voice over the RTP/UDP/IP protocol stack. The data transmission trouble is caused by UDP when the service is provided in private networks and some ISP-provided Internet access networks in the private address space. The Internet telephony users in such networks cannot listen the voices of the other parties in the public Internet or PSTN. Making the problem more difficult, the Internet telephony service considered in this paper gets the incoming voice packets of every session through only one UDP port number. In this paper, three schemes including the terminal proxy, the gateway proxy, and the protocol translation are suggested to solve the problems. The design and implementation of the NAT proxy server based on gateway proxy scheme are described in detail.

Survey on Out-Of-Domain Detection for Dialog Systems (대화시스템 미지원 도메인 검출에 관한 조사)

  • Jeong, Young-Seob;Kim, Young-Min
    • Journal of Convergence for Information Technology
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    • v.9 no.9
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    • pp.1-12
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    • 2019
  • A dialog system becomes a new way of communication between human and computer. The dialog system takes human voice as an input, and gives a proper response in voice or perform an action. Although there are several well-known products of dialog system (e.g., Amazon Echo, Naver Wave), they commonly suffer from a problem of out-of-domain utterances. If it poorly detects out-of-domain utterances, then it will significantly harm the user satisfactory. There have been some studies aimed at solving this problem, but it is still necessary to study about this intensively. In this paper, we give an overview of the previous studies of out-of-domain detection in terms of three point of view: dataset, feature, and method. As there were relatively smaller studies of this topic due to the lack of datasets, we believe that the most important next research step is to construct and share a large dataset for dialog system, and thereafter try state-of-the-art techniques upon the dataset.