• MALSORI
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• no.63
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• pp.139-151
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• 2007

As the Internet becomes prevalent in our lives, harmful contents, such as phornographic videos, have been increasing on the Internet, which has become a very serious problem. To prevent such an event, there are many filtering systems mainly based on the keyword-or image-based methods. The main purpose of this paper is to devise a system that classifies pornographic videos based on the audio information. We use the mel-cepstrum modulation energy (MCME) which is a modulation energy calculated on the time trajectory of the mel-frequency cepstral coefficients (MFCC) as well as the MFCC as the feature vector. For the classifier, we use the well-known Gaussian mixture model (GMM). The experimental results showed that the proposed system effectively classified 98.3% of pornographic data and 99.8% of non-pornographic data. We expect the proposed method can be applied to the more accurate classification system which uses both video and audio information.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.269-275
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• 2003

In this paper, we proposed the robust watermark diminishing distortion of the original data inserting the watermark in specific coefficients of the frequency domain. In case the alpha is more than 0.5. we found that proposed watermark is detected by experiment of MP3, FFT, Cropping and Echo attack. Our proposed method improved the Cox's method in the SNR aspect.

• Journal of the Korean Institute of Telematics and Electronics
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• v.13 no.6
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• pp.12-15
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• 1976

This paper presents the realization method of 90$^{\circ}$ constant phase shifter in the VHF band, constructed by the entirely differens idea from the conventional method of the realization of constant phase shifter in the audio frequency range. The construction of 90$^{\circ}$ constant phase shifter is. simple and can be easily realized by using the distributed constant element, transmission line, and the lumped constant elements, R,C,L.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.709-712
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• 1988

Ultrasonic Doppler Diagnostic System utilizes the Doppler effect for measurement of blood velocity. The sign of the Doppler frequency shift represents blood flow direction. CW(Continuous-Wave) Doppler System uses quadrature detection and phase rotation method to produce simultaneous independent audio and velocity signals for forward and reverse blood flow direction in the time-domain, had been fabricated. But time-domain analyzing such as audio evaluation and zero- crossing detection for instantaneous and mean frequnecy measurement do not provide both an accurate and quantitative result. Therefore, it is necessary to adopt frequency-domain technique to improve system performance. In this paper, we describe a unit which is composed of CW Doppler System and real-time spectrum analyzer (installed TMS 32010 DSP Chip). This unit shows time-dependent spectrum variation and mean velocity of Blood signal.

• Speech Sciences
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• v.8 no.4
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• pp.75-86
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• 2001

This study explored the effectiveness of a hierarchical treatment of aphasic perseveration (TAP) program in a Korean client with transcortical sensory aphasia. The subject with 52% perserveration score (Korean version of Boston Naming Test : K-BNT) was 44 year-old female with MCA (Middle cerebral artery) infarction. The experimental design used was an alternating treatment design with the hierarchical TAP and conventional audio-visual stimulation. The frequency of occurrence of perseverative behaviors and correct response in naming performance were analyzed and compared. It was claimed that the hierarchical TAP was more effective in naming performance than conventional audio-visual stimulation in terms of correct naming response. The frequency of occurrence of perseverative behaviors was lower in hierarchical TAP but the difference was relatively small. Unlike in English, sentence completion task was no longer stimulable while unison speech was very stimulable among the specific strategies of TAP program in Korean. Therefore, it could be said that TAP is language-dependent.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.10A
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• pp.1499-1511
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• 2000

We have designed and implemented a receiver of Eureka-147 Digital Audio Broadcasting (DAB) scheme. DAB used OFDM (orthogonal frequency division multiplexing) as its transmission technique that is very sensitive to synchronization error. So reliable synchronization algorithms have been proposed and evaluated. With the algorithms we proposed a PC based real-time DAB receiver has been implemented. We find that the receiver performs satisfactorily to meet the specifications we determined.

• Journal of Information Processing Systems
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• v.15 no.4
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• pp.890-903
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• 2019

The single carrier frequency domain equalization (SC-FDE) technology is an important part of the broadband wireless access communication system, which can effectively combat the frequency selective fading in the wireless channel. In SC-FDE communication system, the accuracy of timing synchronization directly affects the performance of the SC-FDE system. In this paper, on the basis of Schmidl timing synchronization algorithm a timing synchronization algorithm suitable for FPGA (field programmable gate array) implementation is proposed. In the FPGA implementation of the timing synchronization algorithm, the sliding window accumulation, quantization processing and amplitude reduction techniques are adopted to reduce the complexity in the implementation of FPGA. The simulation results show that the algorithm can effectively realize the timing synchronization function under the condition of reducing computational complexity and hardware overhead.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8A
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• pp.1169-1178
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• 1999

In this paper, the method to improve audio noise when GMS(Global System of Mobile Communication) is built in HPC(Handheld PC)is provided. The biggest problem with quality of audio is improved by connecting GSM module with wire to the widest erea near by HPC Power Ground to reduce impedance difference, and designing amplifier and earpiece, Result of measurement is satisfied with GSM Acoustic Standard. Standard of GSM Audio is declared in GSM 11.10 ETS 300 607-1. Measuring items corresponds to that standard and B&K Type 6712 is used for measurement. The list of measurement presented in this paper is Sending Sensitivity Frequency Response and SLR, Sending Loudness Rating( SLR level), Receiving Sensitivity Loudness Rating(RLR level), Talker Sidetone(STMR), Stability margin, and Echo Loss(ERL)

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.303-309
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• 2016

This paper proposes a novel approach to the problem of audio signal clustering using a stacked autoencoder. The proposed stacked autoencoder learns an efficient representation for the input signal, enables clustering constituent signals with similar characteristics, and therefore the original sources can be separated based on the clustering results. STFT (Short-Time Fourier Transform) is performed to extract time-frequency spectrum, and rectangular windows at all the possible locations are used as input values to the autoencoder. The outputs at the middle, encoding layer, are used to cluster the rectangular windows and the original sources are separated by the Wiener filters derived from the clustering results. Source separation experiments were carried out in comparison to the conventional NMF (Non-negative Matrix Factorization), and the estimated sources by the proposed method well represent the characteristics of the orignal sources as shown in the time-frequency representation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.874-882
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• 2004

MPEG Audio compression algorithm is the international standard for the digital compression of high quality audio using mechanism of the perceptual coding based on psychoacoustic masking. It is necessary to discuss the constraints on designing of common filter banks for MPEG-2 BC and MPEG-2 AAC decoder system, which is not Down yet, mapping audio signals from the time domain into the frequency domain. In this paper, we present an architecture of common synthesis filter whcih can be used for MPEG-2 BC and MPEG-2 AAC decoder using recursive structure. The proposed algorithm is based on recursive architecture that effectively performs common compulsion.