• Journal of Broadcast Engineering
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• v.21 no.4
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• pp.621-629
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• 2016

This paper suggests a new structure of hybrid DRC to enhance the psychoacoustic sound quality of a conventional multiband DRC. The proposed hybrid DRC consists of two serially cascaded stages. The front stage DRC is multiband, and it compresses input based on RMS level detection, whereas, the back stage DRC is single band, and it regulates input according to peak level detection. The proposed hybrid DRC shows better loudness while suppressing distortion by clipping. The proposed algorithm was verified through MATLAB simulation, and it was implemented using an FPGA board for listening test. The test result showed that the proposed hybrid structure enhances overall psychoacoustic sound quality compared to conventional structures, which is based on only RMS or peak level detection.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.965-974
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• 2007

In this paper, we proposed the feature selection algorithm for multi-class genre classification. In our proposed algorithm, we developed GMM separation score based on Gaussian mixture model for measuring separability between two genres. Additionally, we improved feature subset selection algorithm based on sequential forward selection for multi-class genre classification. Instead of setting criterion as entire genre separability measures, we set criterion as worst genre separability measure for each sequential selection step. In order to assess the performance proposed algorithm, we extracted various features which represent characteristics such as timbre, rhythm, pitch and so on. Then, we investigate classification performance by GMM classifier and k-NN classifier for selected features using conventional algorithm and proposed algorithm. Proposed algorithm showed improved performance in classification accuracy up to 10 percent for classification experiments of low dimension feature vector especially.

• The Journal of the Acoustical Society of Korea
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• v.33 no.6
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• pp.400-406
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• 2014

This paper describes a robust watermark decoding model using a SVM(Support Vector Machine). First, the embedding process is performed inversely for a watermarked signal. And then the watermark is extracted using the proposed model. For SVM training of the proposed model, data are generated that are watermarks extracted from sounds containing watermarks by four different embedding schemes. BER(Bit Error Rate) values of the data are utilized to determine a threshold value employed to create training set. To evaluate the robustness, 14 attacks selected in StirMark, SMDI and STEP2000 benchmarking are applied. Consequently, the proposed model outperformed previous method in PSNR(Peak Signal to Noise Ratio) and BER. It is noticeable that the proposed method achieves BER 1% below in the case of PSNR greater than 10 dB.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.10-15
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• 2001

In this paper, we describe the comparison between the combination of features using a speech and music discrimination, which is classifying between speech and music on audio signals. Audio signals are classified into 3classes (speech, music, speech and music) and 2classes (speech, music). Experiments carried out on three types of feature, Mel-cepstrum, energy, zero-crossings, and try to find a best combination between features to speech and music discrimination. We using a Gaussian Mixture Model (GMM) for discrimination algorithm and combine different features into a single vector prior to modeling the data with a GMM. In 3classes, the best result is achieved using Mel-cepstrum, energy and zero-crossings in a single feature vector (speech: 95.1％, music: 61.9％, speech & music: 55.5％). In 2classes, the best result is achieved using Mel-cepstrum, energy and Mel-cepstrum, energy, zero-crossings in a single feature vector (speech: 98.9％, music: 100％).

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2018.11a
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• pp.186-189
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• 2018

KBS는 차세대 방식인 IP전송 기술을 UHD방송에 적용하기로 결정하고 2017년 9월부터 2018년 3월까지 약 7개월에 걸쳐 ALL-IP UHD부조정실 구축을 완료했다. 이후 약 3개월간의 시뮬레이션 기간을 거쳐, 현재 KBS 1TV '아침마당'과 '무엇이든 물어보세요' 를 생방송으로 제작하고 있고 KBS 2TV '그녀들의 여유만만'을 녹화 제작하고 있다. 본 논문은 UHD비디오신호와 오디오신호를 ALL-IP로 전송하기 위해 참조한 표준기술과 각 파트별 구축 세부내용을 소개한다. 또한 향후 지속적으로 발전할 IP제작 시스템에 대해 효율적인 계획과 대응을 할 수 있도록 구축사례에 대한 경험을 결론으로 논한다.

• The Journal of the Acoustical Society of Korea
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• v.27 no.3
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• pp.111-118
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• 2008

In this paper, we propose a new automatic taxonomy generation algorithm for the audio genre classification. The proposed algorithm automatically generates hierarchical taxonomy based on the estimated classification accuracy at all possible nodes. The estimation of classification accuracy in the proposed algorithm is conducted by applying the training data to classifier using k-fold cross validation. Subsequent classification accuracy is then to be tested at every node which consists of two clusters by applying one-versus-one support vector machine. In order to assess the performance of the proposed algorithm, we extracted various features which represent characteristics such as timbre, rhythm, pitch and so on. Then, we investigated classification performance using the proposed algorithm and previous flat classifiers. The classification accuracy reaches to 89 percent with proposed scheme, which is 5 to 25 percent higher than the previous flat classification methods. Using low-dimensional feature vectors, in particular, it is 10 to 25 percent higher than previous algorithms for classification experiments.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.191-199
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• 2006

This paper presents an efficient pre echo control scheme for MPEG Audio coders based on the psychoacoustic model II (PAM-II). Pre echo control is the final step for the calculation of masking threshold in the PAM II. It is to minimize the spread of quantization error over the processing frame. In the conventional encoders, pre echo is reduced by restricting the estimated masking threshold not to exceed the one obtained in the previous frame. The conventional method performs pre echo control not only for short blocks but also for long blocks, which lowers the masking threshold in long blocks and, in turn, increases the quantization noise level of corresponding blocks. This paper proposes an efficient pre echo control process. The test result shows a mean enhancement of more than 0.4 especially for complex signals on the ITU R 5 point audio impairment scale.

• Journal of Korea Multimedia Society
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• v.11 no.11
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• pp.1480-1490
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• 2008

Design of a linear predictor and matching of an entropy coder is the art of lossless audio coding. In this paper, we use the covariance method and the Choleskey decomposition for calculating linear prediction coefficients instead of the autocorreation method and the Levinson-Durbin recursion. These results are compared to the polynomial predictor. Both of them, the predictor which has small prediction error is selected. For the entropy coding, we use the Golomb-Rice coder using the block-based parameter estimation method and the sequential adaptation method with LOCO-land RLGR. The proposed predictor and the block-based parameter estimation have $2.2879%{\sim}0.3413%$ improved compression ratios compared to FLAC lossless audio coder which use the autocorrelation method and the Levinson-Durbin recursion. The proposed predictor and the LOCO-I adaptation method could improved by $2.2879%{\sim}0.3413%$. But the proposed predictor and the RLGR adaptation method got better results with specific signals.

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.410-418
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• 2012

Parametric stereo(PS) and MPEG surround(MPS) are major spatial audio coding(SAC) tools. In this paper, the problem of the inter-channel correlation(ICC) synthesis in the conventional SAC is analyzed. Conventional methods assume that ambient components mixed to two output channels are anti-phased, while the primary components are assumed to be in-phased. This assumption can cause excessive ambient mixing for a negative-valued ICC. As a remedy to this problem, we propose a new ICC synthesis method based on an assumption that the primary components are anti-phased each other for a negative ICC. The proposed method is also applied to the approximation which works in practice. The performance of the proposed method was evaluated by computer simulations and the subjective listening tests verified that the proposed method is effective in not only headphones but also loudspeakers playback.

• Journal of Digital Convergence
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• v.12 no.3
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• pp.269-275
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• 2014

The DAB system was designed to provide CD quality audio and data services for fixed, portable and mobile applications with the required BER below $10^{-4}$. However for the T-DMB system with the video service of MPEG-4 stream, BER should go down $10^{-8}$ by adding FEC blocks which consist of the Reed-Solomon (RS) encoder/decoder and convolutional interleaver/deinterleaver. In this paper we propose two types of turbo coded T-DMB system without altering the puncturing procedure and puncturing vectors defined in the standard T-DMB system for compatibility. One(Type 1) can replace the existing RS code, convolutional interleaver and RCPC code by a turbo code and the other one (Type 2) can substitute the existing RCPC code by a turbo code. Simulation results show that two new turbo coded systems are able to yield considerable performance gain after just 2 iterations. Type 2 system is better than type 1 but the amount of performance improvement is small.