• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.5
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• pp.91-97
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• 2012

Recently, the performances of electrical and electronic devices are improving while the sizes are becoming smaller. As sound-generating systems, the two-dimensional speakers have been developed in place of conventional three-dimensional ones. Piezoelectric materials show the mechanical vibrations due to the voltage applied from outside the materials. The early film speakers had a limitations of output power in that it was not easy to make the conducting macromolecular films on the surfaces of the materials due to the internal chemical properties of materials. We have adopted the corona surface treatment in order to improve the output characteristics by increasing the adhesion of the coating material on to the surface of the center material of piezo film. The results showed the improvement of output power in the wider range of operating frequencies.

• Journal of the Korean Institute of Intelligent Systems
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• v.24 no.3
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• pp.316-322
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• 2014

A movie recommendation system is proposed to learn a preference model of a viewer by using multimodal features of a video content and their evoked implicit responses of the viewer in synchronized manner. In this system, facial expression, body posture, and physiological signals are measured to estimate the affective states of the viewer, in accordance with the stimuli consisting of low-level and affective features from video, audio, and text streams. Experimental results show that it is possible to predict arousal response, which is measured by electrodermal activity, of a viewer from auditory and text features in a video stimuli, for estimating interestingness on the video.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.6
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• pp.1-5
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• 2011

We present the Multi-band orthogonal frequency division multiplexing ultra-wideband (MB-OFDM UWB) technology for increasing the transmission reach of wireless speaker systems. The proposed scheme adopts the Reed-Solomon coding for preventing the random error perfectly and shows the SNR gain in low bit error rate (BER) especially. So, we can increase the maximum reach of MB-OFDM UWB technology since the receiver sensitivity is improved. The simulation environment includes most effects of realistic channel environments such as Additive White Gaussian Noise (AWGN), CM1 channel model, Sampling frequency offset (SFO), Carrier frequency offset (CFO) to improve the simulation accuracy. The simulation results show that the proposed scheme can give a maximum 2 dB SNR gain and increase the transmission reach up to 12.6m.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.439-451
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• 2021

We propose a deep learning-based beamformer combined with spectral subtraction for continuous speech recognition operating in noisy environments. Conventional beamforming systems were mostly evaluated by using pre-segmented audio signals which were typically generated by mixing speech and noise continuously on a computer. However, since speech utterances are sparsely uttered along the time axis in real environments, conventional beamforming systems degrade in case when noise-only signals without speech are input. To alleviate this drawback, we combine online beamforming algorithm and spectral subtraction. We construct a Continuous Speech Enhancement (CSE) evaluation set to evaluate the online beamforming algorithm in noisy environments. The evaluation set is built by mixing sparsely-occurring speech utterances of the CHiME3 evaluation set and continuously-played CHiME3 background noise and background music of MUSDB. Using a Kaldi-based toolkit and Google web speech recognizer as a speech recognition back-end, we confirm that the proposed online beamforming algorithm with spectral subtraction shows better performance than the baseline online algorithm.

• Journal of Broadcast Engineering
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• v.27 no.6
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• pp.914-922
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• 2022

In this paper, we construct a commercial 4K UHD(Ultra High Definition) streaming device that utilizes a 5G mobile communication network as a transport channel and conduct a streaming performance test. It uses RTP(Realtime Transport Protocol) which has transmission quality monitoring capability as a transmission protocol to apply adaptive streaming. In addition, it provides the function to adjust the encoding rate of the video signal so that encoding can be optimized for the change in the bandwidth of the transmission channel. Through the performance test, it is confirmed that the H.265 encoding rate for 4K UHD signal is 48.69Mbps, the average glass-to-glass delay time is 293.60ms, and the average time difference between video and audio for lip sync is 120ms. With the result of performance test, it is shown that the streaming device is applied to 4K UHD Streaming device over 5G mobile communication network.

• Journal of Korea Society of Industrial Information Systems
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• v.18 no.5
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• pp.45-50
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• 2013

Recently, as the market for SMART TV expands, the camera is embedded for providing various user experience. However, this leads to occurrence of camera failure due to TV power up sequence problem, which are usually not detectable in conventional test equipments. Although the failure-detection can be possible by re-generating control signals for audio interface with new equipment, it is expensive and also requires much time to test. In this paper, for SMART TV, FPGA(Field Programmable Gate Array)-based failure-detection system is proposed which can lead to reduction of both cost and time for test.

• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.756-762
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• 2009

The interest of show industry as leisure and culture bring about quantitative increase of public facility. It indicate the effort of bring culture benefits to the disable and the general who want detail information. Especially, there is less technology and service to help them. So it is very necessary method which is using the audial and visual transfer technology to solve the limitation. This study aimed at indicating the method that applied the audial and visual transfer technology which are useful for both The disable and General Audience.

• Journal of KIISE:Software and Applications
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• v.26 no.1
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• pp.52-65
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• 1999

이미지나 비디오, 오디오와 같이 멀티미디어 데이터들은 기존의 단순한 텍스트 기반의 데이터에 비하여 대용량적인 특성과 비정형적인 특성을 가지고 있어서 검색시 많은 어려움이 따른다. 본 논문에서는 대규모의 이미지 데이터베이스에서 효율적이고 신속하게 사용자가 원하는 이미지를 검색할수 있는 내용 기반 검색 시스템을 제시한다. 이를 위해서 본 논문에서는 최근 여러 장점으로 인하여 신호 분석이나 이미지 압축 분야에 많이 사용되는 웨이브릿 변환을 이용하여 이미지 데이터로부터 내용 기반 검색에 사용되는 특징 벡터를 효율적으로 추출하는 기법과 유사성 측정 방법을 제안한다. 그리고, 이러한 특징 추출방법과 유사성 측정 방법을 이용하여 내용 기반 질의 및 검색을 수행할 경우, 검색 조건을 만족하는 객체인데 실수로 검색해내지 못하는 경우인 false dismissals 이 발생하지 않음을 보인다. 또한 대규모 이미지 데이터베이스에서 신속한 내용 기반 검색을 지원하기 위하여 고차원 데이터에 대한 효율적인 색인을 제공하는 X-tree를 이용한 이미지 색인 방법을 보이며 이것이 기존의 순차 검색이나 R*-tree를 이용한 색인 방법보다 신속하게 이미지 데이터들을 검색할 수 있다는 것을 다양한 실험을 통해 보인다. 마지막으로 QBIC에서 제안한 검색 적합성 측정 방법을 이용하여 본 논문에서 제안하는 내용 기반 이미지 검색시스템의 검색 적합성을 보인다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.7
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• pp.90-97
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• 1989

In this paper, we have taken the formation of two-dimension codeword named doubly-encoded code using the Reed-Solomon code, C1(32, 28) with minimum distance 5 and C2(32, 26) with minimum distance 7 and we have had computer simulation of these error correcting processes using modeled R-DAT (Rotationary Digital Audio Tape). As the result, the error rate per symbol has been decreased about 0.05 and on these processes, the newly developed digital signal processing technology such as erro correction using Berlekamp-Massey algorithm in frequency domain have been proven.