• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.229-232
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• 2000

AAL2 has been adopted for bandwidth-efficient trans-mission of low bit tate traffic over ATM networks in ITUT and ATM Forum. Since ATM/AAL2 is expected to be used as a switching technology in third-generation mobile access networks and mobile data traffic is expected to increase rapidly in near future, there must be a need for efficient scheduling scheme satisfying the QoS requirement of ow bit rate voice as well as the one of high bit rate data. In this paper, we propose a new class-scheduling scheme to improve data packet loss probability, while Qos of voice traffic is guaranteed, when data traffic is multiplexed together with mobile voice traffic into a single ATM VCC. The proposed scheme can efficiently support data traffic by assigning a time threshold value to voice traffic. Through simulation study, we show that the proposed scheme does not only achieve better efficiency for providing both mobile voice and data services than HOL class-scheduling scheme and normal FIFO scheme, but also guarantees mean voice packet delay under a certain criteria.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.567-570
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• 2001

In this paper, we proposed flash memory design method for image and voice recoder based on the standard imageㆍvoice codec algorithm. For implementation of this method we designed image voice browser which is application system of flash memory and card using GDS30C6001 USB controller. To process image and voice data we designed root directory of image and voice files repectively. To extend application of image and voice data we added various information to the system.

• Journal of KIISE:Information Networking
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• v.27 no.1
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• pp.68-75
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• 2000

In this paper, a media access control protocol is proposed for voice/data integrated services in the packet CDMA network, and the performance of the proposed protocol is analyzed. The proposed protocol uses the spreading code sensing and the reservation schemes. This protocol gives higher priority to the delay-sensitive voice traffic than to the data traffic. A voice terminal can reserve an available spreading code during a talkspurt to transmit multiple voice packets. On the other hand, whenever a data packet is generated, the data terminal transmits the packet through one of the available spreading codes that are not used by the voice terminals. In this protocol, the voice packets do not come into collision with the data packets. The numerical results show that this protocol can increase the maximum number of voice terminals. The performance for the data traffic degrades by increasing the voice traffic load because of the low priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.22 no.1
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• pp.57-65
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• 2012

People leave voicemails or record phone conversations in their daily cell phone use. Sometimes important voice data is deleted by the user accidently, or purposely to cover up criminal activity. In these cases, deleted voice data must be able to be recovered for forensics, since the voice data can be used as evidence in a criminal case. Because cell phones store data that is easily fragmented in flash memory, voice data recovery is very difficult. However, if there are identifiable patterns for the deleted voice data, we can recover a significant amount of it by researching images of it. There are several types of voice data, such as QCP, AMR, MP4, etc.. This study researches the data recovery solutions for EVRC codec and AMR codec in QCP file, Qualcumm's voice data format in cell phone.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.2
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• pp.171-181
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• 1988

In this study, a packet voice/data terminal(PVDT) that services both voice and data in the packet-switched network is implemented. The software structure of the PVDT is designed according to the OSI 7 layer architecture. The discrimination of voice and data is made in the link layer. Voice packets have priority over data packets in order to minimize the transmission delay, and are serviced by a simple protocol so that the overhead arising form the retransmission of packets may be minimized. The hardware structure of the PVDT is divided into five modules; a master control module, a speech proessing module, a speech activity detection module, a telephone interface module, and an input/output interface module. In addition to the hardware implementation, the optimal reconstruction delay of voice packets to reduce the influence of delay variance is analyzed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.5
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• pp.916-922
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• 2001

A medium access control protocol is proposed for integrated voice and data services in the packet CDMA network with a small coverage. Uplink channels are composed of time slots and multiple spreading codes for each slot. This protocol gives higher access priority to the delay-sensitive voice traffic than to the data traffic. During a talkspurt, voice terminals reserve a spreading code to transmit multiple voice packets. On the other hand, whenever generating a data packet, data terminals transmit a packet based on the status Information of spreading codes in the current slot, which is received from base station. In this protocol, voice packet does not come into collision with data packet. Therefore, this protocol can increase the maximum number of voice terminals.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.2
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• pp.137-143
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• 1997

From now on, the network is being developed into PSTN(public switched telephone network) and PDN(public data network), that is depend on the form of data. The former one pursues sending voice, and the latter one pursues sending data. But it causes big loss of the economy and efficiency. So, ISDN, processing voice and data at same time, gives a big profit to user. To enlarge the ISDN at the narrow area, it is necessary that study to send the mixture form of voice and data in LAN environment. So, this paper proposes the algorithm about the mixture form of voice and data in ethernet and token-ring. that is widely used in these days.

• Proceedings of the KSR Conference
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• 2008.06a
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• pp.1962-1967
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• 2008

A event recorder system stores a train speed and the related information for train operation in real time. Using those information, we can analysis the train operation and the reason of train accident. Currently the event recorder only manipulate the data related the train operation mechanically and electrically. In this paper we propose the event recorder to record the voice communication between the manager in the control center and train operator. By recording the voice communication in the high speed train, the correctness of analysis of train accident can be increased. The system architecture of the event recorder with voice recording is studied and interface between other equipment is proposed. And the software architecture of new event recorder is developed. We study the method of converting analog voice signal into digital data and compressing method. Also the architecture of memory to store the compressed voice data and regeneration of original analog voice are studied.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.2
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• pp.57-66
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• 2010

In this paper, we propose an enhanced voice recognition algorithm using adaptive MFCC(Mel Frequency Cepstral Coefficients) and neural network. Though it is very important to extract voice data from the raw data to enhance the voice recognition ratio, conventional algorithms are subject to deteriorating voice data when they eliminate noise within special frequency band. Differently from the conventional MFCC, the proposed algorithm imposed bigger weights to some specified frequency regions and unoverlapped filterbank to enhance the recognition ratio without deteriorating voice data. In simulation results, the proposed algorithm shows better performance comparing with MFCC since it is robust to variation of the environment.