• Proceedings of the Korea Information Processing Society Conference
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• 2013.05a
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• pp.303-306
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• 2013

실제공간체감을 극대화하기 위해 실제 물리적인 공간과 가상현실 공간을 융합하는 인터랙티브 아키텍쳐 기반 적응형 혼합현실 기술이 최근 연구되고 있다. 이러한 혼합현실 공간에서 동적인 사용자 위치에 따라 물리공간적 몰입감 증대를 위한 오디오 Sweet Spot 최적화 기술을 연구하였다. 이를 위해 주파수 대역 별 소리의 물리적 감쇠현상을 활용하여 주파수 별 오디오 신호 보상 전처리를 통해 동적인 사용자 위치에 원음과 동일한 음색의 오디오 Sweet Spot이 형성이 가능한지 실험한 결과 주파수 별 감쇠의 차이를 보정함으로써 원음 그대로의 음색이 재현될 수 있다는 것을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.25 no.2
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• pp.56-60
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• 2006

Based on the fact that audio signals in the time domain have the generalized Gaussian distribution. an optimal parameter estimation of the ML (maximum likelihood) test based audio watermark decoder. which leads to the minimal bit error rate, is Proposed. Its superiority of performance over the existing estimation and the conventional correlation based decoder is demonstrated experimentally.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.2
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• pp.63-68
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• 2015

This paper proposes a robust multiple audio watermarking algorithm using improved QIM(quantization index modulation) with adaptive stepsize for different signal power and SVM(support vector machine) decoding model. The proposed algorithm embeds watermarks into both frequency magnitude response and frequency phase response using QIM. This multiple embedding method can achieve a complementary robustness. The SVM decoding model can improve detection rate when it is not sure whether the extracted data are the watermarks or not. To evaluate robustness, 11 attacks are employed. Consequently, the proposed algorithm outperforms previous multiple watermarking algorithm, which is identical to the proposed one but without SVM decoding model, in PSNR and BER. It is noticeable that the proposed algorithm achieves improvements of maximum PSNR 7dB and BER 10%.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.6
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• pp.251-257
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• 2014

An actuator system is a type of motor designed to control a mechanism operated by a source of energy, in the form of an electric current by converting energy into some kind of motion. As audio actuators, transforming electric voltage signal into audio signal, speakers and amplifiers are commonly used. In applications of industry, high output power systems are required. For these systems to generate high-quality output, it is essential to control output impedance of audio systems. We have developed an adaptable power supply for driving active amplifier systems with variable loads. Depending on the changing values of resistance of the speaker which produces audible sound by transforming electric voltage signal, the power supply source of the active amplifier can generate the maximum power delivered to the speaker by an adaptable change of loads. The amplifier is well protected from the abrupt increment of peak current and an excess of current flow.

• Journal of Korea Multimedia Society
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• v.6 no.5
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• pp.816-823
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• 2003

In this paper, we propose a blind audio watermarking using the tonal characteristic. First, we explain the perceptional effect of tonal on the existed researches and shout the experimental result that tonal characteristic is more stable than other characteristics used in previous watermarking studies against several signal processing. On the base of the result, we propose the blind audio watermarking using the relation among the signals on the frequency domain which compose a tonal masker. To evaluate the sound quality of our watermarked audios, we used the SDG(Subjective Diff-Grades) and got the average SDG 0.27. This result says the watermarking using the perceptional effect of tonal is available from the viewpoint of non-perception. And we detected the watermark hits from the watermarked audios which were changed by several signal processing and the detection ratios with exception of the time shift processing were over 98%. About the time shift processing, we applied the new method that searched the most proper position on the time domain and then detected the watermark bits by the ratio of 90%.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.06a
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• pp.86-89
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• 2016

본 논문에서는 멀티미디어 콘텐츠의 명료도를 향상시켜 사용자가 주변 환경과 관계없이 안정적인 볼륨에서 오디오 청취를 할 수 있는 다이얼로그 명료도 향상 알고리즘을 제안한다. 최근 모바일 기기들의 발달로 다양한 환경에서 영화, TV 또는 동영상 등의 멀티미디어 콘텐츠를 즐기는 일이 늘어나고 있다. 이러한 경우 시청자는 주변 환경에 따라 영상의 오디오 볼륨을 조절하게 되는데 주변 소음에 비하여 과하게 증폭된 볼륨은 주변에 피해를 끼치거나 고막에 손상을 일으킬 수 있으며 반대로 주변에 비해 너무 작은 오디오 볼륨은 시청을 어렵게 한다는 단점이 있다. 본 논문에서는 수신단에서 멀티미디어 콘텐츠의 오디오 신호로부터 다이얼로그 성분을 검출하여 음성 명료도 향상 알고리즘을 적용시켜 동일한 볼륨에서도 음성의 명료도를 높이는 알고리즘을 제안한다. 본 알고리즘은 다이얼로그를 검출하여 단순히 증폭 시키는 기존 기술들과 달리 전체 에너지는 유지하면서 명료도에 중요한 영향을 미치는 주파수 대역에 에너지를 집중시키는 에너지 재분배 방식을 이용해 동일한 볼륨에서도 더 높은 음성 명료도를 기대할 수 있다. 컴퓨터 시뮬레이션을 통해 본 논문에서 제안한 알고리즘이 명료도에 중요한 영향을 미치는 주파수대역을 적절히 증폭시킴을 확인할 수 있었다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.63-68
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• 2016

Audio mixer is an electronic device which performs a mixing of multiple audio signals. Digital mixer having various functions and scalability is spreaded thanks to advanced DSP and IT technology. However, digital mixer is more vulnerable to stability comparing to conventional analog mixer in the digital error or software error sense because its control is executed by SW. To solve this problem, in this paper, we propose a multi-channel digital analog hybrid mixer scheme, digital mixer error detection mechanism and malfunctioning switching technique. Also we develop the audio mixer having digital-analog hybrid structure. By simulation, we can sense the error of digital mixer except power loss in a 120ms, change into analog mixer mode automatically and provide continuous broadcasting function without mixer function loss.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.81-88
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• 2010

In this paper, we proposed an automatic music/non-music signal discrimination system from broadcasting audio signal as a preliminary study of building a sound source monitoring system in real broadcasting environment. By reflecting human speech articulation characteristics, we used three simple time-domain features such as energy standard deviation, log energy standard deviation and log energy mean. Based on the experimental threshold values of each feature, we developed a rule-based algorithm to classify music portion of the input audio signal. For the verification of the proposed algorithm, actual FM broadcasting signal was recorded for 24 hours and used as source input audio signal. From the experimental results, the proposed system can effectively recognize music section with the accuracy of 96% and non-music section with that of 87%, where the performance is good enough to be used as a pre-process module for the a sound source monitoring system.