• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.821-835
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• 1998

The Digital Audio-Visual Council(DAVIC) that was established to develop useful multimedia communication services has completed the specifications for providing on-demand services such as Movie on Demand(MoD), Teleshopping and accepting Internet service. And then they are proceeding the works to suport converstional communcation services like Plain Old Telecphone Service(POTS), Video telephone, Video teleconferencing. In this paper, we prpose an efficient terminal architecture which can provide conversational multimedia communication services on DAVIC Set-Top Box (STB) environments. To apply the implemented conversational terminal to the multipoint communication environment, we considered the factors of Qurlity of Services(QoS) that determine grade of conversational communication service. We also present the inter-working scheme and that system structure to satisfy QoS by using new MPEG video bridge which gurantees end to end delay requirements as major element of QoS for achieving the real time communication and does not accompany visual quality degradation.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• Journal of Digital Convergence
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• v.14 no.8
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• pp.233-243
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• 2016

Due to speech recognition problems in noisy environment, Audio Visual Speech Recognition (AVSR) system, which combines speech information and visual information, has been proposed since the mid-1990s,. and lip reading have played significant role in the AVSR System. This study aims to enhance recognition rate of utterance word using only lip shape detection for efficient AVSR system. After preprocessing for lip region detection, Convolution Neural Network (CNN) techniques are applied for utterance period detection and lip shape feature vector extraction, and Hidden Markov Models (HMMs) are then used for the recognition. As a result, the utterance period detection results show 91% of success rates, which are higher performance than general threshold methods. In the lip reading recognition, while user-dependent experiment records 88.5%, user-independent experiment shows 80.2% of recognition rates, which are improved results compared to the previous studies.

• Journal of IKEEE
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• v.16 no.4
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• pp.395-404
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• 2012

In this paper, the perceived loudness metering algorithm based on ITU-R BS.1387-1 was investigated and implemented, and its performance was evaluated by applying to 23 pure tones and 9 digital audio samples. Error of the tone test results compared with ISO226:2003 was below 5%, and sample test results, in comparison with Moore's algorithm, showed deviation of less than 4.7% and correlation of 0.96. On the other hand, it was investigated how the implemented algorithm's performance was subject to auditory pitch scale. Its result showed that the algorithm with 37 auditory filters, through correcting a bias effect, has a good performance of less than 2% in comparison with the one with 109 auditory filters.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.1
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• pp.42-49
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• 2003

OFDM should be used for the fourth generation communication for high speed communication. Because of high spectral efficiency and high tolerance to fading channel, OFDM is applied to many high speed wire and wireless communication such as DAB (Digital Audio Broadcast), DVB(Digital Video Broadcast), IMT 2000 etc. Inter-modulation, however, is derived from PAR(Peak to Average Power Ratio) of OFDM signals. The paper describes PTS(Partial Transmit Sequence) and SLM(Select Mapping) of an existing methods which can reduce PAR. And then this papers proposed the new method that is called "Combine method". The method proposed in this paper is to combine PTS and SLM. As a result of the simulation, Combine PAR method is better than the existing methods.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.400-405
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• 2006

In this paper, we present effective methods for music summarization which summarize music automatically. It could be used for sample music of on-line digital music provider or some music retrieval technology. When summarizing music, we use different two methods according to music length. First method is for finding sabi or chorus part of music which can be regarded as the most important part of music and the second method is for extracting several parts which are in different structure or have different mood in the music. Our proposed music summarization system is better than conventional system when structure of target music is explicit. The proposed method could generate just one important segment of music or several segments which have different mood in the music. Thus, this scheme will be effective for summarizing music in several applications such as online music streaming service and sample music for Tcommerce.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.965-969
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• 2003

In this paper, the implementation technique of digital crossover network using perfect linear phase IIR filters is presented. This system has various advantages which cannot be obtained from analog crossover network such as linear phase response, flat group delay and sharp cut-off at low-order over audio frequency band. The simulation results show that the group delay response is maximally flat and twice more attenuation in stop-band than the prototype elliptic IIR filter at all desired frequency.