• Journal of the Institute of Electronics Engineers of Korea TE
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• 제39권3호
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• pp.21-27
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• 2002

The digital watermarking methods have been proposed as a solution for solving the illegal copying and proof of ownership problems in the context of multimedia data. But it is still difficult to have been overcame the problem of the protection of property to multimedia data, such as digital images, digital video, and digital audio. This paper describes a watermarking algorithm that embeds non-linearly audio logo watermark data which is converted from audio signal of the ownership in the components of pixel intensities in an original image and that insists of ownership by hearing the audio signal transformed from the extracted audio logo through the speaker. Experimental results show that our algorithm using audio logo proposed in this paper is robust against attacks such as particularly lossy JPEG image compression. 

• Journal of Broadcast Engineering
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• 제6권1호
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

• IEIE Transactions on Smart Processing and Computing
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• 제2권1호
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• Journal of Broadcast Engineering
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• 제19권6호
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• pp.819-825
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• 2014

In this paper, monaural ambient component extraction algorithm based on nonnegative matrix factorization (NMF) is described. The ambience component extraction algorithm in this paper is developed for audio upmixing system; Recent researches have shown that they can enhance listener envelopment if the extracted ambient signal is applied into the multichannel audio upmixing system. However, the conventional method stores all of the audio signal and processes all at once, so it cannot be applied to streaming system and digital signal processor (DSP) system. In this paper, the ambient component extraction algorithm based on on-line nonnegative matrix factorization is developed and evaluated to solve the problem. As a result of analysis of the processed signal with spectral flatness measures in the experiment, it was shown that the developed system can extract the ambient signal similarly with the conventional batch process system.

• Journal of Korea Multimedia Society
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• 제21권9호
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• pp.1068-1075
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• 2018

We propose a method for driving loudspeakers from class D audio power amplifiers in current mode, instead of in conventional voltage mode, which was impossible with the feedback circuitry. Unlike analog audio amplifiers, Class D audio power amplifiers have signal delay between the input and output signals, which makes it difficult to apply the feedback circuitry for current-mode driving. The idea of the pre-distortion scheme used for the compensation of the non-linearity of RF power amplifiers is adapted to remedy the impedance variation effect of the loudspeakers for current driving. The method uses the speaker model for the pre-distorter to compensate for the speaker impedance variation with frequency. The simulation and test results confirms the validity of the proposed method.

• ETRI Journal
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• 제24권3호
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• pp.181-189
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• 2002

Digital watermark technology is now drawing attention as a new method of protecting digital content from unauthorized copying. This paper presents a novel audio watermarking algorithm to protect against unauthorized copying of digital audio. The proposed watermarking scheme includes a psychoacoustic model of MPEG audio coding to ensure that the watermarking does not affect the quality of the original sound. After embedding the watermark, our scheme extracts copyright information without access to the original signal by using a whitening procedure for linear prediction filtering before correlation. Experimental results show that our watermarking scheme is robust against common signal processing attacks and it introduces no audible distortion after watermark insertion.

• Proceedings of the IEEK Conference
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.241-242
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• 2006

MPEG Surround is an audio coding technology that represents multi-channel audio signal with downmixed audio signal(s) and very low bitrate side information based on Binaural Cue Coding. The side information consists of Inter-Channel Level Difference, Inter-Channel Correlation, and payloads. These two parameters are correspondent to the well-known spatial parameters in psycho-acoustics, Inter-aural Level Difference (ILD) and Inter-Aural Cross Correlation (IACC). Though ICLD is to provide perceptually equivalent ILD to the listener, however, the ILD of the original multi-channel audio signal and that of the MPEG Surround encoded signal was different. The difference between two ILD values is defined as ILD Distortion (ILDD). This paper provides how ILDD can be applied to enhance sound quality in MPEG Surround and how much ILDD is decreased.

• Journal of the Institute of Convergence Signal Processing
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• 제15권1호
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• pp.1-8
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• 2014

Digital watermarking techniques are attracting attention as a proper solution to protect copyright for multimedia data. This paper proposes a new audio watermarking method based on Discrete Cosine Transformation (DCT) and Discrete Wavelet Transformation (DWT) for copyright protection. In our proposed watermarking method, the original audio is transformed into DCT domain and divided into two parts. Synchronization code is applied on the signal in first part and 2 levels DWT domain is applied on the signal in second part. The absolute value of DWT coefficient is divided into arbitrary number of segments and calculates the energy of each segment and middle peak. Watermarks are then embedded into each middle peak. Watermarks are extracted by performing the inverse operation of watermark embedding process. Experimental results show that the hidden watermark data is robust to re-sampling, low-pass filtering, re-quantization, MP3 compression, cropping, echo addition, delay, and pitch shifting, amplitude change. Performance analysis of the proposed scheme shows low error probability rates.

• Journal of the Institute of Electronics Engineers of Korea CI
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• 제40권5호
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• pp.289-297
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• 2003

The performance of a bimodal system is affected by the accuracy of the endpoint detection from the input signal as well as the performance of the speech recognition or lipreading system. In this paper, we propose the endpoint detection method which detects the endpoints from the audio and video signal respectively and utilizes the signal to-noise ratio (SNR) estimated from the input audio signal to select the reliable endpoints to the acoustic noise. In other words, the endpoints are detected from the audio signal under the high SNR and from the video signal under the low SNR. Experimental results show that the bimodal system using the proposed endpoint detector achieves satisfactory recognition rates, especially when the acoustic environment is quite noisy.

• Journal of the Institute of Electronics Engineers of Korea SP
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• 제47권5호
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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.