• ETRI Journal
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• v.31 no.1
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• pp.65-67
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• 2009

We introduce an advanced terrestrial digital multimedia broadcasting (AT-DMB) system that overcomes the limitation of data transmission rates of T-DMB by doubling it with the same frequency bandwidth. In this letter, we propose an efficient algorithm which generates a scalable transport stream in AT-DMB by multiplexing certain types of elementary streams encoded using scalable video coding and an MPEG-surround audio coder for high-quality multimedia services.

• ETRI Journal
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• v.39 no.2
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• pp.145-150
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• 2017

As wearable devices are powered by batteries, they need to consume as little energy as possible. To address this challenge, in this article, we propose a synergistic technique for energy-efficient approximate speech signal processing (ASSP) for wearable devices. More specifically, to enable the efficient trade-off between energy consumption and sound quality, we synergistically integrate an approximate multiplier and a successive approximate register analog-to-digital converter using our enhanced conversion algorithm. The proposed ASSP technique provides ~40% lower energy consumption with ~5% higher sound quality than a traditional one that optimizes only the bit width of SSP.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.35C no.1
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• pp.34-41
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• 1998

In this paper, I designed a P1394 serial bus chip as new bus interface architecture which can transmit the multimedia data at the rate of 400 Mbps and guarantee necessary bandwidth. because multimedia data become meaningless data after appropriate time, it is necessary to transfer multimedia data in real time, P1394 serial bus chip designed in this paper support isochronous transfer mode to solve this problem. Also, designed P1394 serial bus chip can transfer high quality video data or high quality audio data because it support the speed of 400 Mbps. While user must set device ID manually in previous interface such as SCSI, device ID is automatically determined if user connect each node with designed P1394 serial bus cable and power on. To design this chip, I verified the behavioral of the entrire system and synthesized layout. Also, I did layout the analog blocks and blocks which must be optimized in full custom.

• Proceedings of the IEEK Conference
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• 2002.06e
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• pp.83-86
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• 2002

In this paper while a router is routing all packet to the next hop, it inspects whether there is congestion on this current hop router or not and if the router discovers that it has some congestion, it informs that the packet is experienced to congestion. The packet arrived to next hop including some information about the congestion is processed first and it has wider bandwidth than another packet The amount of congestion is recorded to the DS field of IP header by congestion experience level. In the next hop when the packet including the congestion information is routed, the standard packet dropping ratio of the current router is changed in proportion to congestion experience that is recorded in IP header on of that. When the packet that has experienced congestion before is arrived, the router extends the drop threshold value not to drop the packet. It mean that transferring the audio or video stream, if the packet is already experienced the congestion in another hop, the router can provide the better service quality about 15∼25% than another.

• The Journal of the Korea institute of electronic communication sciences
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• v.3 no.4
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• pp.228-232
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• 2008

For the past several years, many countries have passed or have recommended legislation making it illegal to use in-hand mobile phones while driving and several manufacturers have released car kit solutions enabling hands-free operation of the mobile phone. But an automobile environment can pose extremely harsh physical conditions impacting audio quality, safety, and reliability. This article introduced a Car Kits that provided a total entertainment and telematics solution, which support all current features within the constraints of low power consumption, form factor, price, ease-of-use, manufacture ability, testability and high total quality.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.353-354
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• 2006

This paper suggests the packet loss recovery method to communicate in real-time in the Internet. To reduce the effects of packet loss, Forward Error Correction(FEC) that adds redundant information to voice packets can be used. The major cause for speech quality degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme is evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.695-698
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• 2000

The services such as real-time audio and video applications have become increasingly popular, especially over the Internet. Furthermore, as being commercialized those contents on the Internet require quality of service (QoS) support to ensure their performance. PPP is the best solution to of for those kinds of services. The reason why we want to employ PPP is this satisfies most of the requirements associated with remote connectivity to an NSP, such as IP address assignment, security, and AAA (authentication, authorization and accounting). In addition, since ISPs and corporations are familiar with PPP based connectivity, easy migration from existing ISP infrastructure is expected, if QoS is guaranteed. But so for PPP has had no field to ensure the quality of service. This article presents the solution by using some tunneling protocols and the draft [1] that proposed additional LCP option fields to negotiate QoS. To communicate each other, after negotiating those option fields, over various protocols such as ATM, Ethernet, and etc. tunneling protocol is used. Following sections will mention those briefly. And the service provision to offer the end-to-end communication with negotiated QoS will also be proposed.

• Journal of Communications and Networks
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• v.4 no.3
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• pp.199-208
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• 2002

The downlink rate scheduling problem is considered for CDMA networks with multiple users carrying packets of heterogeneous traffic (voice/audio only, bursty data only or mixed traffic), with each type having its own distinct quality of service requirements. Several rate scheduling algorithms are developed, the common factor of which is that part of the decision on which users to serve is based on a function of the deadline of their head-ofline packets. An approach of Andrews et al., in which the basic Earliest-Deadline-First algorithm is studied for similar systems, is extended to result in better performance by considering a more efficient power usage and by allowing service of more than one user per timeslot if the power resources permit it. Finally, the performance of the proposed schemes is compared through simulations.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.77-85
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• 2001

This paper proposes a method of time-scale modification of polyphonic audio signals based on a sinusoidal model. The signals are modeled with sinusoidal component and noise component. A multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in time-scale modification a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. For extracting sinusoidal components and calculating their parameters matching pursuit algorithm is applied to each analysis frame of subband signal. In accordance with spectrum analysis a psychoacoustic model implementing the effect of frequency masking is incorporated with matching pursuit to provide a resonable stop condition of iteration and reduce the number of sinusoids. The noise component obtained by subtracting the synthesized signal with sinusoidal components from the original signal is modeled by line-segment model of short time spectrum envelope. For various polyphonic audio signals the result of simulation shows suggested sinusoidal modeling can synthesize original signal without loss of perceptual quality and do more robust and high quality time-scale modification for large scale factor because of representing transients without any perceptual loss.