• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.327-330
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• 2008

This paper shows an implementation of WLAN-based Audio/Video(A/V) system for multimedia home networks. Proposed WLAN-based A/V system can transmit multimedia data of high quality. The entire system consists of a 16-bit RISC controller, a program ROM, a SRAM, timers, an interrupt controller, a DART, GPIOs, an I2C and the OFDM modem supporting for the IEEE 802.11g standard. The simple MAC functions are implemented by firmware on an embedded 16-bit RISC controller. The OFDM modem supports a complete set of data rates up to 54Mbps. Proposed the system is implemented by an Altera FPGA EP1S60F1020C6 device, a 10-bit 2-ch DAC, a 10-bit 2-ch ADC and RF/IF chips.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.257-261
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• 2008

The Digital Radio Mondiale (DRM) system is a digital broadcasting standard designed for use in the LF, MF and HF bands of the broadcasting bands below 30 MHz. The system provides both superior audio quality and improved user services / operability compared with existing AM transmissions. In this paper, we propose a variable point Prime Factor FFT design method for Digital Radio Mondiale (DRM) system. Proposed method processes a various size IFFT/FFT of Robustness Mode on DRM standard efficiently by composing Radix-Prime Factor FFT Processing Unit of form similar to Radix-4 by insertion of a variable Prime Factor Twiddle Factor and Garbage data. So, we improved limitation that cannot process 112/176/256/288 FFT of each mode of DRM system with a existent Radix Processor and increase memory size and memory access time for IFFT/FFT processing by software processing in case of implementation with a existent high speed DSP.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.317-322
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• 2004

In this paper, It was describe the processes of development of HiFi speaker system. Woofer and tweeter were fabricated by unskilled students and their 1.5 parameters were identified by known mass method. Based on T-S parameters port enclosure was designed and built by means of software. Acoustic radiation phenomena of port enclosure were simulated and compared to test result. Acoustic pressure difference between lower frequency and higher frequency was flattened by adopting optimal crossover network. Finally, built HiFi speaker system was showed good sound quality and sound pressure and electrical impedance was well agreed with test results each other.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.338-341
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• 2003

최근의 인터넷 음악방송은 MP3 오디오를 기반으로 하는 TCP 프로토콜의 전송방식이 일반적이다. TCP방식의 전송은 HTTP 프로토콜을 이용한 파일 전송방식으로 네트워크의 부하가 급증할 경우 TCP의 특성으로 인해 음악의 끊김 현상이 발생하여 QoS 문제가 발생한다. 본 논문은 실시간 전송방식의 RTP(Real-time Transfer Protocol) 프로토콜을 이용하여 MP3 오디오 기반의 생방송 시스템 개발에 대한 연구로서, 기존의 TCP 방식의 음악의 끊김 현상을 개선하기 위한 모듈 구현을 목적으로 한다. 본 연구에서는 MP3 오디오 전송에 따른 QoS(Quality Of Service) 개선을 위하여 인터리빙 기법을 이용한 SPH/RPH(Send Payload Handler/ Receive Payload Handler) 모듈을 구현하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2018.06a
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• pp.53-54
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• 2018

본 논문에서는 스펙트로그램을 이용하여 딥 러닝 기반으로 오디오 장르와 품질의 다중 정보를 동시에 분류하는 기술을 제안한다. 기존 딥 러닝 기반의 오디오 정보 인식 기술은 각각의 정보 인식을 목표로 독립 네트워크를 설계하고, 여러 정보를 동시에 인식하기 위하여 각각에 특화된 여러 네트워크를 사용한다. 이러한 문제점을 보완하기 위해 본 논문에서는 디지털 오디오의 대표 특성인 스펙트로그램을 기반으로 범용성이 있는 특성을 추출하고, 단일 네트워크로 학습시켜 장르 및 품질을 동시에 분류하는 다중 분류 기술을 제안한다. 제안하는 방법으로 단일 분류 성능과 유사한 다중 분류 성능을 얻을 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4B
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• pp.349-359
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• 2009

In this paper, we propose a stereo-video transmission method for reduction of delay and maximization of 3D effect. Conventional multimedia synchronization algorithms were designed to achieve minimum delay and synchronize multiple video and audio streams, however, they could not be effective for 3D video transmission. In this paper, we proposed a synchronization algorithm by considering the minimum error of time difference between streams for 3D effect. The minimum error of time difference for 3D effect was derived based on a 3D subjective quality test. We compute display time of the delivered videos within the allowed time-difference and the video are displayed according to the display time. To evaluate the performance of the proposed algorithm, we implemented a real-time video communication system and subjective quality test has been conducted with the proposed system. We found that video quality displayed by the proposed system. We found that video quality displayed by the proposed algorithm ranks 'good' and 'excellent' in the DMOS (Differential Mean Opinion Score) scale, based on the MOS (Mean Opinion Score) test.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.822-829
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• 2009

This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

• Journal of the Korea Convergence Society
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• v.9 no.10
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• pp.303-313
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• 2018

In order to provide quality services across international airports, airline personnel must rapidly and effectively develop and share knowledge. Combining components of adaptive structuration theory (AST) and media synchronicity theory (MST), a research framework was developed to convey three distinct stages of knowledge sharing. We use the grounded theory research method for the qualitative data collected from audio transcripts of employees learning how to use and work with company issued smartphones with push-to-talk functionalities. Data was collected from 33 operations personnel. The results of the content analysis are recorded for the elements of each of the three concepts of our research framework. During the social interaction stage, the content of the audio conversations shifts mainly from conflict management to task management; for media synchronicity, from quality to quantity; for productive outcomes, from efficiency to commitment. New insights are uncovered from our analysis of data from the field as users advance from learning how to use the mobile devices, to using the devices for managing knowledge for their work in the airline industry.

• Journal of Broadcast Engineering
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• v.21 no.6
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• pp.889-898
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• 2016

Virtual Bass System (VBS) is widely used to extend the lower frequency limit of small loudspeakers, which generates harmonics of a fundamental frequency. The perceptual quality of the VBS is highly dependent on the harmonic weighting strategy. There have been several weighting methods, including exponential attenuation and timbre matching. However, it is essential to match phases between harmonics in the original signal and generate harmonics to precisely convey the weighting strategy. This paper shows the limitations of the previous harmonic weighting schemes and proposes a new harmonic weighting scheme. The proposed weighting scheme proposes phase matching between the original and generated harmonics and varies the slope of the attenuation weighting dynamically according to the missing fundamental frequency. Objective and subjective tests show that the proposed harmonic weighting scheme provides more natural and effective bass perception in a limited situation than the conventional schemes, which implies that the phase matching is essential for the high quality bass enhancement.

• Journal of Broadcast Engineering
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• v.17 no.3
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• pp.529-539
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• 2012

Testing is an important part of quality control in the software life cycle. One of the most important issues in the software testing is to generate the appropriate test cases. Generally, the software can be tested based on product understanding. However, it is not easy to create test cases that can ensure the quality of the software according to the clients' request. Especially, it is difficult to create test cases for abnormal or exceptional situations. In this paper, we present a novel approach to generate exceptional test cases at the design level of UML model. Furthermore, we describe the results of actual experiment where DAB(Digital Audio Broadcasting) parsing program is tested with previous method and also with the proposed method. The results implies that our proposed method of generating test cases for exceptional situations detect more faults than that of previous method by 7.08%.