• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.181-190
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• 2002

This paper proposes a motion-video multimedia system needed for computer applications like remote lecturing, distance learning, product demonstrations, and so on. The applications need a multimedia system which can author and play a motion-Video that is composed with computer screen images and audio continuously varing as time flows, not with real motion videos. Since the computer screen images are not like the real world video images in several rejects, MPEG is not competent as a compression algorithm for computer screen images raring continuously In this paper a new compression algorithm has been proposed, and a multimedia system that authors and plays a motion-video file which contains computer screen images and audio has been implemented. Also this paper shows the result of performance evaluation of both the compression algorithm and the multimedia system implemented in the paper.

• Proceedings of the IEEK Conference
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• 2003.11b
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• pp.243-246
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• 2003

In this paper, we propose a new digital audio watermarking technique with the wavelet transform. The watermark is embedded by eliminating unnecessary information of audio signal based on human auditory system (HAS). This algorithm is an audio watermarking method, which does not require any original audio information in watermark extraction process. In this paper, the masking effect is used for audio watermarking, that is, post-tempera] masking effect. We construct the window with the synchronization signal and we extract the best frame in the window by using the zero-crossing rate (ZCR) and the energy of the audio signal. The watermark may be extracted by using the correlation of the watermark signal and the portion of the frame. Experimental results show good robustness against MPEG1-layer3 compression and other common signal processing manipulations. All the attacks are made after the D/A/D conversion.

• Journal of Broadcast Engineering
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• v.1 no.1
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• pp.67-73
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• 1996

In general, the high quality audio coding(HQAC) has the structure of the convertional data compression techniques combined with moodels of human perception. The primary auditory characteristic applied to HQAC is the masking effect in the spectral domain. Therefore spectral techniques such as the subband coding or the transform coding are widely used[1][2]. However no effort has yet been made to apply the temporal masking effect and temporal redundancy removing method in HQAC. The audio data compression method proposed in this paper eliminates statistical and perceptual redundancies in both temporal and spectral domain. Transformed audio signal is divided into packets, which consist of 6 frames. A packet contains 1536 samples($256{\times}6$) :nd redundancies in packet reside in both temporal and spectral domain. Both redundancies are elminated at the same time in each packet. The psychoacoustic model has been improved to give more delicate results by taking into account temporal masking as well as fine spectral masking. For quantization, each packet is divided into subblocks designed to have an analogy with the nonlinear critical bands and to reflect the temporal auditory characteristics. Consequently, high quality of reconstructed audio is conserved at low bit-rates.

• ETRI Journal
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• v.27 no.3
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• pp.330-333
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• 2005

The Inter-Channel Level Difference (ICLD) is a cue parameter to estimate spectral information in a binaural cue coding that has been recently in the spotlight as a multichannel audio signal compression technique. Even though the ICLD is an essential parameter, it is generally distorted by quantization. In this paper, a new modified ICLE representation method to minimize the quantization distortion is proposed by adopting a flexible determination of the reference channel and the unidirectional quantization. Our experimental result confirms that the proposed method improves the multichannel audio output quality even with the reduced bit-rate.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.160-163
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• 2002

This paper presents a statistical technique for audio watermarking. We describe the application of the promising public key watermarking method to the patchwork algorithm. Its detection process does not need the original content nor the secret key used in the embedding process. Special attention is given to statistical method working in the frequency domain. We will present a solution of robust watermarking of audio data. In this scheme, an extension of patchwork audio watermarking is presented which enables public detection of the watermark. Experimental results show good robustness of the approach against MP3 compression and other common signal processing manipulations.

• 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.

• Journal of Digital Contents Society
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• v.14 no.2
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• pp.199-206
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• 2013

Various methods of data compression have been developed to handle data within limited storage capacity and limited transmission speed. Recompression technology, a technology most recent among them, is a technology that can embed data regardless of the information entropy of a data. Recompression technology separates original multimedia data in to blocks and embeds 0 or 1 according to whether each block is flipped or not. In this paper, this technology has been applied on audio files. And was able to implement reversible watermarking for audio files.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.3
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• pp.87-95
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• 2010

Portable multi-media products which can service the highest audio-quality by using lossless audio codec has been released and the international lossless codecs, MPEG-4 audio lossless coding(ALS) and MPEG-4 scalable lossless coding(SLS), were standardized by MPEG in 2006. The simple profile of MPEG-4 ALS, it supports up to stereo, was defined by MPEG in 2009. The lossless audio codec should have low-complexity in stereo to be widely used in portable multi-media products. But the previous researches of MPEG-4 ALS have focused on an improvement of compression ratio, a complexity reduction in multi-channels coding, and a selection of linear prediction coefficients(LPCs) order. In this paper, the complexity and compression ratio of MPEG-4 ALS encoder is analyzed in simple profile of MPEG-4 ALS, the method to reduce a complexity of MPEG-4 ALS encoder is proposed. Based on an analysis of complexity of MPEG-4 ALS encoder, the complexity of short-term prediction filter of MPEG-4 ALS encoder is reduced by using the low-complexity filter that is proposed in previous research to reduce the complexity of MPEG-4 ALS decoder. Also, we propose a joint coding decision method, it reduces the complexity and keeps the compression ratio of MPEG-4 ALS encoder. In proposed method, the operation of joint coding is decided based on the relation between cross-correlation of residual and compression ratio of joint coding. The performance of MPEG-4 ALS encoder that has the method and low-complexity filter is evaluated by using the MPEG-4 ALS conformance test file and normal music files. The complexity of MPEG-4 ALS encoder is reduced by about 24% by comparing with MPEG-4 ALS reference encoder, while the compression ratio by the proposed method is comparable to MPEG-4 ALS reference encoder.

• Proceedings of the IEEK Conference
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• 2000.06c
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• pp.93-96
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• 2000

Currently, the most important technology is the compression methods in the multimedia society. Audio files are rapidly propagated through internet. MP-3 is offered to CD tone quality in 128Kbps, but 64Kbps below tone quality is abruptly down and high bitrate. on the other hand, MPEG-II AAC (Advanced Audio Coding) is not compatible with MPEG-I, but AAC has a high compression ratio 1.4 better than MP-3. Especially, AAC has max. 7.1 channel and 96KHz sampling rate. In this paper, the perceptual model is dealt with 44.1KHz sampling rate for SMR(Signal to Masking Ratio)