• Title/Summary/Keyword: Voice transmission

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Reservation and Status Sensing Multiple Access Protocol in Slotted CDMA Systems

  • Lim, In-Taek;Ryu, Young-Tae
    • Journal of information and communication convergence engineering
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    • v.8 no.5
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    • pp.513-518
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    • 2010
  • This paper proposes a medium access control protocol for integrated voice and data services in slotted CDMA systems. The proposed protocol, which is named as RCSSMA (Reservation Code and Status Sensing Multiple Access), adopts a code reservation and status sensing schemes. RCSSMA protocol gives higher access priority to the voice traffic than data traffic for reducing the packet dropping probability. The voice terminal reserves an available spreading code to transmit voice packets during a talkspurt, whereas the data terminal transmits a packet over one of the available spreading codes that are not reserved by the voice terminals. In this protocol, the voice packets never contend with the data packets. Packet dropping probability and average data packet transmission delay are analyzed using a Markov chain model.

Real-Time Travelling Control of Mobile Robot by Conversation Function Based on Voice Command (대화기능에 의한 모바일로봇의 실시간 주행제어)

  • Shim, Byoung-Kyun;Lee, Woo-Song;Han, Sung-Hyun
    • Journal of the Korean Society of Industry Convergence
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    • v.16 no.4
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    • pp.127-132
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    • 2013
  • We describe a research about remote control of mobile robot based on voice command in this paper. Through real-time remote control and wireless network capabilities of an unmanned remote-control experiments and Home Security / exercise with an unmanned robot, remote control and voice recognition and voice transmission are possible to transmit on a PC using a microphone to control a robot to pinpoint of the source. Speech recognition can be controlled robot by using a remote control. In this research, speech recognition speed and direction of self-driving robot were controlled by a wireless remote control in order to verify the performance of mobile robot with two drives.

Wireless Communication Real-Time Travelling Control of Mobile Robot by Voice Command (음성명령에 의한 모바일로봇의 무선통신 실시간 주행제어)

  • Shim, Byoung-Kyun;Han, Sung-Hyun
    • Journal of the Korean Society of Manufacturing Process Engineers
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    • v.10 no.6
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    • pp.33-38
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    • 2011
  • We describe a research about remote control of mobile robot based on voice command in this paper. Through real-time remote control and wireless network capabilities of an unmanned remote-control experiments and Home Security / exercise with an unmanned robot, remote control and voice recognition and voice transmission are possible to transmit on a PC using a microphone to control a robot to pinpoint of the source. Speech recognition can be controlled robot by using a remote control. In this research, speech recognition speed and direction of self-driving robot were controlled by a wireless remote control in order to verify the performance of mobile robot with two drives.

Transmission Performance of VoIP Traffics on Underwater MANET (수중 MANET에서 VoIP 트래픽의 전송 성능)

  • Kim, Young-Dong
    • The Journal of the Korea institute of electronic communication sciences
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    • v.11 no.12
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    • pp.1187-1192
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    • 2016
  • Performance analysis results are limited to of network level, because network level transmission parameters are used for performance measure and analysis of network design, construction and operation on underwater MANET, With this way of performance analysis based on network level, it is not easy to analyze transmission performance related with user level transmission quality. In this paper, transmission performance focused on application traffic be required by user is investigated to supplement weakness of performance analysis based on network level. Voice traffic, which is expected to be increasingly used on underwater MANET, is considered as application service, Some conditions for underwater MANET will be proposed to support transmission quality, MOS, CCR and EED, etc.. A computer simulation based on NS-2 is used for performance measure, voice traffic is generated as VoIP specification.

A Study for the Voice channel extension method using Code Division Multiplexing (부호분할 다중화 기법을 이용한 음성 회선 확대 방안 연구)

  • 권기형;신용조
    • Journal of the Korea Society of Computer and Information
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    • v.3 no.4
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    • pp.103-109
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    • 1998
  • Domastic telephony transmission networks mainly using El in 2.048Mb㎰ is composed to 30 channels and each channel is assigned to 64Kb㎰ voice coding rate. El method always uses TDM, so it is fixed channels. In this paper, it shows that using CDM enlarge the subscribers and voice channels

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A Study on Improvement Effect of voice information transmission using Auralization at the hydraulic turbine dynamo room in Dam (가청화를 이용한 댐 수차 발전기실의 음성정보전달 개선효과에 관한 연구)

  • Kook, Joung-Hun;Ju, Duck-Hoon;Jung, Eun-Jung;Kim, Jae-Soo
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2007.11a
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    • pp.263-267
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    • 2007
  • Even though Waterpower Generation as pollution-free has its own merit of contribution by supply of good quality electricity, due to the noise made at the time of its operation, a normal mutual communication among the workers and technicians engaging at the hydraulic turbine dynamo room is almost impossible, and since those finishing materials had been used mainly by reflection material, it is actual situation that when working for maintenance in the hydraulic turbine dynamo room, as counterpart's voice vibrates, its working efficiency is difficult to ensure. On such view point, this Research has conducted Psycho-acoustics Experiment about voice Definition using Auralizational Technique, on the object for the hydraulic turbine dynamo room that improved its acoustic performance by computer simulation. As the result of Study, it was known that the clearness of sound with regard to voice information transmission was apparently improved in all items than before improvement. Therefore, it is considering that these results would be utilized usefully when renovation on the hydraulic turbine dynamo room in the future.

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A Study on the Improvements of the Speech Quality by using Distribution Characteristics of LSP parameters in the EVRC(Enhanced Variable Rate Codec) (LSP 파라미터의 분포특성을 이용한 EVRC의 음질개선에 관한 연구)

  • Min, So-Yeon;Na, Deok-Su
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.12 no.12
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    • pp.5843-5848
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    • 2011
  • To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC's basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.

A Consideration on the Minimum Transmission Loss for the Intraoffice Call Path Based on the Listener Echo (수화자 반향을 고려한 자국내 최소 전송손실에 대한 고찰)

  • Jang, Chung-Ryong;Hong, Jin-Woo
    • Proceedings of the KIEE Conference
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    • 1987.07b
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    • pp.894-897
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    • 1987
  • Listener echos, which arise in multiple 4-wire loop connections(MLC) during the evolving switched telephone network, impare voice-band data signal transmission performance. This paper first shows the calculation method of the total number of listener echo loops over N 4-wire physical loops and presents the additative law for listener echos. It next demonstrates that about 4 dB should be ensured to Eke the transmission loss of intraoffice call path be minimum for the voice-band data service in a digital local switch.

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Performance Comparison of CDMA and TDMA protocols in radio access system for Integrated Voice/Data Services (음성 및 데이터서비스를 위한 무선접속시스템에서 CDMA와 TDMA방식의 성능비교)

  • 고종하;양영님;이정규
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.6A
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    • pp.820-831
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    • 1999
  • In this paper, we have compared the performance of a D-TDMA protocol with that of a CDMA protocol, in radio access system for integrated voice/data services.The D-TDMA protocol is based on a generic dynamic channel assignment approach to be followed a combination of “circuit mode” reservation for voice calls, along with dynamic first-come-first served assignment of remaining capacity for data messages. In the CDMA protocol, we have used the voice activity detection to reduce the interface power of other mobiles in internal and external cells, and analyzed the interference power ratio. Also we have computed BER(Bit Error Rate) by using this interference power ratio and evaluated voice blocking probability(voice packet loss probability) and data transmission delay, according to average data length and average data arrival rate.We have found the CDMA protocol achieves comparatively higher performance for short data length, regardless of data arrival rate. Otherwise, the data transmission delay of D-TDMA protocol is shorter than that of the CDMA protocol for long data message.

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Feasibility Study of IEEE 802.15.4 LR-WPAN to the Real-time Voice Application (IEEE 802.15.4 LR-WPAN의 실시간 음성 데이터 응용에 대한 적용 가능성 연구)

  • Hur, Yun-Kang;Kim, You-Jin;Huh, Jae-Doo
    • IEMEK Journal of Embedded Systems and Applications
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    • v.2 no.2
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    • pp.82-94
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    • 2007
  • Wireless sensor networking technology is one of the basic infrastructures for ubiquitous environment. It enables us to gather various sensory data such as temperature, humidity, gas leakage, and speed from the remote sensor devices. To support these networking functions, IEEE WPAN working group makes standards for PHY and MAC, while ZigBee Alliance defines the standards for the network, security, and applications. The low-rate WPAN was emerged to have the characteristics of network resilience, low cost, and low power consumption. It has a broad range of applications including, but not limit to industrial control and monitoring, home automation, disaster forecast and monitoring, health care. In order to provide more intelligent and robust services, users want voice-based solutions to accommodate to low-rate WPAN. In this paper, we have evaluated voice quality of an IEEE 802.15.4 standard compliant voice node. Specifically, it includes the design of a voice node and experiments based on the prediction of voice quality using the E-model suggested by ITU-T G.107, and the network communication mechanisms considering beacon-enabled and nonbeacon-enabled networks for real-time voice communications.

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