• The Journal of the Korea Contents Association
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• v.12 no.6
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• pp.78-91
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• 2012

The Terrestrial DTV service compliant with ATSC has been advancing for years. In KBA(Korean Broadcasters Association), a multi-channel service was broadcasted on air during the period of the 2006 FIFA World Cup Germany with the various type of MMS(Multi Mode Service) using MPEG-2 encoding method. MMS Service can provides not only one HD channel but also serveral additional services within 6MHz bandwidth. Using digital video compression technology(MPEG-2), many various programs such as HDTV, SDTV, Audio and Data are able to be transmitted within the same bandwidth. From November 2009, KBS has been preparing an advanced MMS service, 'Korea-View' which has both methods of encoding, MPEG-2 and H.264 that is compliant ATSC mobile standard, A/153. Korea-View is a kind of multi-channel broadcast service to provide one HD and 3 SD programs with the bandwidth of 6MHz. Terrestrial multi-channel service is required to focuse on expanding viewer service. Such Terrestrial multi-channel services will contribute to transferring to digital broadcasting and to extending the viewers' welfare. Due to advances in digital technology, Pay-TV channels has increased to hundreds. Even though digital switchover is being proceeded, terrestrial broadcasters have been unable to deliver multi-channel services. In this paper, technical features and differences of MMS and Koreaview will be analyzed regarding terrestrial multi-channel broadcasting services, and the politic direction will be proposed in accordance with introduction of future service.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.127-134
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• 2009

In spite of controversial debates on aesthetic issues of computer-generated electronic music, rapid advancement of music technologies in the past decade have resulted proliferation of using virtual software synthesizers and samplers in music composition. Computer-based music production platform has become not only a norm among some of contemporary music composers but also vital apparatus for their compositional process. There are two imperative parts of this compositional process involving sampling in computer-based music production, which are commercially available sample libraries that include pre-recorded audio samples, and music production software that processes them. The purpose of this study is to investigate the process and effectiveness of reconstructive compositional process utilizing distinctive features of sampling on computer music production software. This study addresses issues such as: the definition of audio sampling, how sampling is incorporated in compositional process, and what features of music production software are particularly effective in various musical expressions. The result of this study will hopefully accommodate and fulfill the needs of electronic and acoustic musicians' creativeness.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.7
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• pp.823-832
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• 2016

This paper presents a generation method of test streams for verifying conformance of an UHD broadcasting receiver including decoders for video and audio as well as parsers for PSIP and closed caption data. The proposed test streams for video/audio signals can evaluate conformance of HEVC, AC-3 and DTS-HD standards. Especially, test streams for HEVC video compression standard can be used for testing syntax compliance and error resilience for a HEVC decoder. Moreover, the proposed test streams for system/program and closed caption can be applied for verifying parsers for PSIP and CEA-708 standards.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.2
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• pp.27-35
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• 2016

A Low jitter dual output frequency synthesizer for smart audio devices is described in this paper. It has been fabricated in a 1.8 V Dongbu $0.18-{\mu}m$ CMOS process. Output frequency is controlled by 3 rd order Sigma-Delta Modulation and digital divider. The frequency synthesizer has a size of $0.6mm^2$, frequency range of 0.6-200 MHz, loop bandwidth of 350 kHz, and rms jitter of 11.4 ps-21.6 ps.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.15 no.5
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• pp.335-342
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• 2022

In this paper, a single-bit 3rd-order feedforward delta sigma modulator is proposed for audio applications. The proposed modulator is based on a class-C inverter for low voltage and power applications. For the high-precision requirement, the class-C inverter with regulated cascode structure increases its DC gain and acts as a low-voltage subthreshold amplifier. The proposed Class-C inverter-based modulator is designed and simulated in 180-nm CMOS process. With no performance loss and a low supply voltage compatibility, the proposed class-C inverter-based switched-capacitor modulator achieves high power efficiency. This design achieves an signal-to-noise-and-distortion ratio (SNDR) of 93.9 dB, an signal-to-noise ratio (SNR) of 108 dB, an spurious-free dynamic range (SFDR) of 102 dB, and a dynamic range (DR) of 102 dB at a signal bandwidth of 20 kHz and a sampling frequency of 4 MHz, while only using 280 μW of power consumption from a 0.8-V power supply.

• Journal of the Korea Society of Computer and Information
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• v.29 no.4
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• pp.47-54
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• 2024

The importance of cultural arts education utilizing digital tools is increasing in terms of enhancing tech literacy, self-expression, and developing convergent capabilities. The creation process and evaluation of innovative multi-modal AI, provides expanded creative audio-visual experiences in users. In particular, the process of creating music with AI provides innovative experiences in all areas, from musical ideas to improving lyrics, editing and variations. In this study, we attempted to empirically analyze the process of performing tasks using an Audio and Music Generative AI platform and discussing with fellow learners. As a result, 12 services and 10 types of evaluation criteria were collected through voluntary participation, and divided into usage patterns and purposes. The academic, technological, and policy implications were presented for AI-powered liberal arts education with learners' perspectives.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.1
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• pp.49-55
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• 2003

The use of digital filters in the signal process field is increasing rapidly with development of the modern industrial society. Especially, detail processors, Y/C separators, ghost removing filters, standard converters (NTSC to PAL or PAL to NTSC) and noise reducers, all of which use digital filters, tend to be used in digital video and audio processing, CATV and various communication fields. Generally, there are two different digital filters, the Rf (infinite impulse response) filter and the FIR (finite impulse response) filter in digital filter. In this paper, we have designed FIR filter which has the phase linearity and the easiness of creation. In the design of the FIR digital filter, the window function is used to alleviate the ripples caused by Gibbs Phenomenon around the cut off frequency of the band pass. But there're some problems to choose proper window function for the design destination due to its fixed values. Therefore, in this paper, we designed a modified Hanning window with new parameter which is adaptively chosen corresponding to design objectives. The digital filter was simulated to prove the validity of the model and it was compared with the Hamming, the Manning, the Blacknan and the Kaiser window function. And we have used peak side-lobe and transient characteristics as standard of judgement.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.31-40
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• 2005

In this paper, we develop an approach toward JSC(Joint Source-Channel Coding) method for MPEG-4 based FGS(Fine Granular Scalability) video coding and transmission in fixed and mobile receiving environment(Digital Audio Broadcasting, DAB). The source coder used MPEG-4 FGS video codec, the channel coder used RCPC(Rate Compatible Punctured Convolution) code and the modulation method used QPSK modulation. We have considered channel environment of AWGN and mobile receiving environment. This study determined optimum Trade-off point between source bit rate and channel coding rate in variable channel states. We compared FGS-JSC method and general single layer CBR(Constant Bit Rate) transmission. In this results, FGS-JSC was appeared better performance than CBR transmission.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.11
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• pp.19-29
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• 2001

The Pan-European Digital Audio Broadcasting(DAH) system's performance is characterized and improved with the aid of turbo codec. From the fact that the first bit among the four coded bits at the RCPC coding defined in the Eureka 147 DAD system is not. punctured and always transmitted, this paper proposes a new turbo coded DAB system model that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the Rician fading channel and the Rayleigh fading channel in conjunction with DAD transmission mode I and III suitable for the terrestrial single frequency network and satellite broadcasting.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.509-517
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• 2013

In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.