• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.4
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• pp.892-898
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• 2017

This paper describes the development of AVN(Audio Video Navigation) software for vehicle information analysis and gesture recognition. The module that examine the CAN(Controller Area Network) data of vehicle in the designed software analyzes the driving state. Using classified information, the AVN software converge vehicle information and hand gesture information. As the result, the derived data is used to match the service step and to perform the service. The designed AVN software was implemented in HW platform that common used in vehicles. And we confirmed the operation of vehicle analysing module and gesture recognition in a simulated environment that is similar with real world.

• ETRI Journal
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• v.19 no.4
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• pp.344-362
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• 1997

The delay which is one of the quality of service parameters is considered to be a crucial factor for the effective usage of real-time audio and video streams in interactive multimedia collaborations. Among the various causes of the delay, we focus in this paper on the local delay concerned with the schemes which handle continuous inflow of encoded data from constant or variable bit-rate audio and video encoders. We introduce two kinds of implementation approaches, pull model and push model. While the pull model periodically pumps out the incoming data from the system buffer, the push model receives events from the device drivers. From our experiments based on Windows NT 3.51, it is shown that the push model outperforms the other for both constant and variable bit-rate streams in terms of the local delay, when the system suffers reasonable loads. We interpret this experimental data with M/G/1 multiple vacation queuing theories, and show that it is consistent with the queuing theoretic interpretations.

• Journal of the Korean Institute of Telematics and Electronics
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• v.12 no.3
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• pp.1-7
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• 1975

In this paper, a method of writing and reproducing high density data with ordinary Audio cassette deck is discribed. In writing, the data N or NRZ code are modulated into PM code to take the positive code N(1) and the negative code N(0) are taken from the complement of the NRZ code, each of which are written into 2 channel track. In reading, the error corrected and the clock pulse can be generated from the reading pulse itself. Also, without modifying the interior circuit of the deck it is possible to use the deck in both the data and audio by adapting a few simple circuits. Over the range of 25HZ-4KHZ, it was possible to write and reproduce at the speed of 787 bps transmission rate.

• Proceedings of the KIEE Conference
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• 2004.07d
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• pp.2429-2431
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• 2004

In this paper, we have designed a remote controlable HDR System using all embedded linux board. The system is composed of three parts - a HDR System, a PC client program for remote control and a Nameserver for registering and aquisition of the IP address. The system is built in an embedded board using a linux kernel. With the Linux the system can support networking and file system for a hard disk management In addition, the system embeds a web-server and a ftp-server for remote manipulation and file transfer. And the hardwares of the system are controlled by the linux device driver mechanism. MPEG1/2 technique is used to compress TV tuner signal and external analog video/audio signal. And compressed data is stored in a hard disk. The data stored in the system is accesable through lan or internet. And RTP protocol is used to enable the system to service live stream of instant video/audio input.

• Journal of IKEEE
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• v.14 no.4
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• pp.297-304
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• 2010

This paper presents a link layer design of DisplayPort interface with a state machine based on packet processing. The DisplayPort link layer provides isochronous video/audio transport service, link service, and device service. The merged video, audio main link, and AUX channel controller are implemented with 7,648 LUTs(Loop Up Tables), 6020 register, and 821,760 of block memory bits synthesized using a FPGA board and it operates at 203.32MHz.

• Proceedings of the Korea Multimedia Society Conference
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• 2004.05a
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• pp.370-373
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• 2004

The rapid proliferation of mobile device such as PDA allows users more ubiquitous access to multimedia information. The user mobility provides users a uniform vision of their preferred working environment independently of their current points of attachment. Supporting the user mobility requires the Player capable of efficiently presenting the multimedia contents. MPEC-4 provides not only the description for coding audio and video (as its predecessors MPEG-1 and MPEG-2), but also for coding images, animations, interactivity and protecting content. With MPEG-4, we present interactive media using multiple objects - audio, video, image, 2D geometry, and text - in a single format. Therefore we propose the MPEC-4 Player for PDA. The proposed MPEG-4 Player for PDA supports mobility, portability and personality.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.3
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• pp.493-497
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• 2005

This paper is about the research of the User Agent implementation under wireless embedded environment, using SIP which is one of protocol components construct the VoIP system. The User Agent is made of the User Agent configuration block, the device thread block to control devices and the SIP stack block to process SIP messages. The device thread consists of the RTP thread and the sound lard device processing block. Futhermore, the SIP stack consist of the worker thread to process proxy events, the SIP transceiver and SIP thread to transfer and receive SIP messages. The H/W platform is a board included the Intel's XScale PXA255 processor, flash memory, SDRAM, Audio CODEC module and wireless LAN threough PCMCIA socket, furthermore a microphone and headphone is used by the audio 1/0. The system has embedded linux kernel 2.4.19. For embedded environment, the function of User Agent and SIP method is diminished. Finally, the resource of system could be reduced about $12.9\%$, compared to overall system resource, by minimizing peripherals control and excepting TCP.

• The KIPS Transactions:PartB
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• v.12B no.6 s.102
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• pp.631-636
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• 2005

Recently, research started to focus on Digital Multimedia Broadcasting(DMB), while the DMB service is already being launched to provide high quality audio, video and data broadcasting in Korea. Therefore, the demand for multimedia contents is also rose very sharply. In this paper, we introduced an MPEG-4 contents authoring system with mobile device for Digital Multimedia Broadcasting. Our system allows both single-handed and collaborative authoring modo. Further more, to overcome the delaytime for decoding the BIFS information, we also introduced an efficient algorithm to generate dynamic and light BIFS information.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.5
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• pp.41-49
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• 2010

Being popularized the use of portable entertainment/information devices, the demand on flash memory has been also increased radically. In general, flash memory reveals various error patterns by the devices it is mounted, and thus the memory makers are trying to minimize error ratio in the final process through not only the electric test but also the data integrity test under the same condition as real application devices. This process is called an application-level memory test. Though currently various flash memory testing devices have been used in the production lines, most of the works related to memory test depend on the sensual abilities of human testers. In case of testing the flash memory for MP3 devices, the human testers are checking if the memory has some errors by hearing the audio played on the memory testing device. The memory testing process like this has become a bottleneck in the flash memory production line. In this paper, we propose an audio comparison technique to support the efficient flash memory test for MP3 devices. The technique proposed in this paper compares the variance change rate between the source binary file and the decoded analog signal and checks automatically if the memory errors are occurred or not.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.5
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• pp.35-41
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• 2008

This paper presents the design method of current-mode signal processing for high speed and low power digital audio signal processing. The digital audio processor requires a digital signal processing such as fast Fourier transform (FFT), which has a problem of large power consumption according to the settled point number and high speed operation. Therefore, a current-mode signal processing with a switched Current (SI) circuit was employed to the digital audio signal processing because a limited battery life should be considered for a low power operation. However, current memory that construct a SI circuit has a problem called clock-feedthrough. In this paper, we examine the connection of dummy MOS that is the common solution of clock-feedthrough and are willing to calculate the relation of width between dummy MOS for a proposal of the design methodology for improvement of current memory. As a result of simulation, in case of that the width of memory MOS is 20um, ratio of input current and bias current is 0.3, the relation of width between switch MOS and dummy MOS is $W_{M4}=1.95W_{M3}+1.2$ for the width of switch MOS is 2~5um, it is $W_{M4}=0.92W_{M3}+6.3$ for the width of switch MOS is 5~10um. Then the defined relation of MOS transistors can be a useful design guidance for a high speed low power digital audio processor.