• The Journal of the Acoustical Society of Korea
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• v.28 no.1E
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• pp.29-34
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• 2009

MPEG4 High Efficiency Advanced Audio Coding (HE-AAC) and MPEG Surround are one of the most efficient combinations for low bit rate multi-channel audio coding. Based on the fact that these two codecs have identical quadrature mirror filter (QMF) analysis and synthesis structures, we propose a dual-domain connection scheme for the codecs. Specifically two time-domain connection methods are analyzed and compared to the QMF subband-domain connection method. Experimental results show that both the time-domain connection methods cause no subjective sound quality degradation compared to the QMF subband-domain connection method, which verifies that one can select either of them depending on application scenarios.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2063-2066
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• 2002

In this paper, we propose a structure far lossless audio coding method. Prediction model is used in the wavelet transform domain. After DWT, wavelet coefficients is quantized and decorrelated by prediction modeling. The DWT can be constructed to critical bands. We can get a lower data rate representation of audio signal which has a good quality like the result of perceptual coding. Then the prediction errors are efficiently coded by the Golomb-coding method. The prediction coefficients are fixed for reducing the computational burden when we find prediction coefficients.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.86-96
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• 2013

Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.369-379
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• 2016

This paper deals with an efficient method for skipping inter-channel phase differences (IPD) in the MPEG surround of the unified speech and audio coding (USAC). Based on the psycho-acoustic sensitivity on the IPD, we estimate a threshold on IPD, below which we can not notice degradation in spatial cue. We propose an IPD skip method, in which any IPDs within the threshold are set to zero and are not transmitted. The proposed IPD skip method gives about 38% savings in terms of bit amount for IPD. Nevertheless, in the MUSHRA test, the proposed method does not show any noticeable degradation in the decoded audio quality.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.4
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• pp.156-163
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• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.580-588
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• 2009

In this paper, we describe technical trends, standard activities, and upcoming issues relating on UHDTV audio, which requires high quality realistic sound. We also propose a proper solution to it for domestic broadcasting and telecommunication environments.

• 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 Korean Institute of Communications and Information Sciences
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• v.33 no.3C
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• pp.240-246
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• 2008

An MPEG-2/4 AAC decoder on 16-bit fixed-point processor is presented in this paper. To meet audio quality criteria, despite small word length, special design methods for 16-bit foxed-point AAC decoder were devised. This paper presents particular algorithms for 16-bit AAC decoding. We have implemented an efficient AAC decoder using the proposed algorithms. Audio contents can be replayed in the decoder without quality degradation.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.712-715
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• 1999

In this paper we propose high quality audio coding algorithm using psychoacoustic modelling and the adaptive wavelet Packet decomposition. The bit allocation scheme exploits the remnants of temporal correlations that exist in the wavelet packet coefficients by SPIHT. The proposed algorithm achieve almost transparent coding of monophonic compact disk(CD) quality signals at about 44 kbps.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.253-260
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• 2004

This paper proposes a novel algorithm for hybrid lossless audio coding, which employs an integer wavelet transform and a linear prediction model. The proposed algorithm divides the input signal into flames of a proper length, decorrelates the framed data using the integer wavelet transform and linear prediction and finally entropy-codes the frame data. In particular, the adaptive Golomb-Rice coding method used for the entropy coding selects an optimal option which gives the best compression efficiency. Since the proposed algorithm uses integer operations, it significantly improves the computation speed in comparison with an algorithm using real or floating-point operations. When the coding algorithm is implemented in hardware, the system complexity as well as the power consumption is remarkably reduced. Finally, because each frame is independently coded and is byte-aligned with respect to the frame header, it is convenient to move, search, and edit the coded, compressed data.