• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.48-55
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• 2000

Recently due to rapid improvement of integrated circuit design environment, size of IC design is to become large to possible design System on Chip(SoC) that one chip with multi function enclosed. Also cause to this rapid change, consumption market is require to spend smallest time for new product development. In this paper to propose a methodology can design a large size IC for save time and applied to design of MPEG Layer 2 decoder to can use audio receiver in digital broadcast system. The digital broadcast audio decoder in this paper is pointed to save hardware size as optimizing algorithm. MPEG Layer 2 decoder algorithm is include MAC to can have an effect on hardware size. So coefficients are using sign digit expression. It is for hardware optimization. If using this method can design MAC without multiplier. The designed audio decoder is using 14,000 gates hardware size and save 22% (4000 gates) hardware usage than using multiplier. Also can design MPEG Layer 2 decoder usable digital broadcast receiver for short time.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.93-96
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• 2000

This paper has been proposed an audio DAC structure composed of FIRs and IIR filters as digital interpolation filter to integrate the off-chip analog low-pass filter on-a-chip. The passband ripple(< 0.41${\times}$fs), passband attenuation(at 0.41${\times}$fs) and stopband attenuation(> 0.59${\times}$fs) of the Δ$\Sigma$ modulator output using the proposed digital interpolation filter had ${\pm}$ 0.001 ［㏈］, -0.0025［㏈］ and -75［㏈］, respectively. Also the inband group delay was 30.07/fs［s］ and the error of group delay was 0.1672%. Also, the attenuation of stopband has been increased -20［㏈］ approximately at 65［㎑］, out-of-band. Therefore the RC products of analog low-pass filter on chip have been decreased compared with the conventional digital interpolation filter structure.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.768-771
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• 2004

In this paper, we propose a content-based audio genre classification algorithm that automatically classifies the query audio into five genres such as Classic, Hiphop, Jazz, Rock, Speech using digital signal processing approach. From the 20 second query audio file, 54 dimensional feature vectors, including Spectral Centroid, Rolloff, Flux, LPC, MFCC, is extracted from each query audio. For the classification algorithm, k-NN, Gaussian, GMM classifier is used. In order to choose optimum features from the 54 dimension feature vectors, SFS (Sequential Forward Selection) method is applied to draw 10 dimension optimum features and these are used for the genre classification algorithm. From the experimental result, we verify the superior performance of the SFS method that provides near $90{\%}$ success rate for the genre classification which means $10{\%}$-$20{\%}$ improvements over the previous methods

• 한국정보통신설비학회:학술대회논문집
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• 2006.08a
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• pp.219-222
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• 2006

In this paper we present a cryptanalysis of the generalized patchwork algorithm under the assumption that the attacker possesses only a single copy of the watermarked audio. In the scheme, watermark is inserted by modifying randomly chosen DCT values in each block of the original audio. Towards the attack we first fit low degree polynomials (which minimize the mean square error) on the data available from each block of the watermarked content. Then we replace the corresponding DCT data of the at-tacked audio by the available data from the polynomials to construct an attacked audio. The technique nullifies the modification achieved during watermark embedding. Experimental results show that recovery of the watermark becomes difficult after the attack.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.34 no.1
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• pp.115-127
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• 2024

The recent increase in digital audio media has greatly expanded the size and diversity of sound data, which has increased the importance of sound data analysis in the digital forensics process. However, the lack of standardized procedures and guidelines for sound data analysis has caused problems with the consistency and reliability of analysis results. The digital environment includes a wide variety of audio formats and recording conditions, but current audio forensic methodologies do not adequately reflect this diversity. Therefore, this study identifies Life-Cycle-based sound data elemental technologies and provides overall guidelines for sound data analysis so that effective analysis can be performed in all situations. Furthermore, the identified elemental technologies were analyzed for use in the development of digital forensic techniques for sound data. To demonstrate the effectiveness of the life-cycle-based sound data elemental technology identification system presented in this study, a case study on the process of developing an emergency retrieval technology based on sound data is presented. Through this case study, we confirmed that the elemental technologies identified based on the Life-Cycle in the process of developing digital forensic technology for sound data ensure the quality and consistency of data analysis and enable efficient sound data analysis.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.237-244
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• 2002

In this paper we mention the optimization of MPEG-4 VM (Moving Picture Expert Group-4 Verification Model) GA (General Audio) AAC (Advanced Audio Coding) encoder and the design of the decoder for PDA (Personal Digital Assistant) using MPEG-4 VM source. We profiled the VMC source and several optimization methods have applied to those selected functions from the profiling. Intel Pentium III 600 MHz PC, which uses windows 98 as OS, takes about 20 times of encoding time compared to input sample running time, with additional options, and about 10 times without any option. Decoding time on PDA was over 35 seconds for the 17 seconds input sample. After optimization, the encoding time has reduced to 50% and the real time decoding has achieved on PDA.

• Journal of Digital Convergence
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• v.12 no.7
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• pp.77-83
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• 2014

Currently, a lot of colleges and universities offer Acoustics and audio engineering courses. In this paper, we analyze and classify the current state of the graduate level curriculums of the area. For the purposes, we focus on graduate school courses of the U.S. and Europe where audio engineering is highly advanced. They were classified into three different types depending on the educational objectives. In addition, the representative cases of each type are presented to examine the characteristics of the subjects.

• Journal of Digital Contents Society
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• v.15 no.3
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• pp.357-363
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• 2014

In this paper, we propose a new audio forensic marking system for protecting the copyright of the Internet-based music streaming services. In the proposed method, in order to guarantee the QoS of the streaming service, high speed, and generates a forensic mark inserted MP3 file. We make pre-marking process and generate a new forensic marked MP3 file, a combination of the pre-marked MP3 frame, the inserted user information. Experimental results show that the proposed method satisfactory results robustness and imperceptibility, and real-time properties. In addition, we were confirmed that the real-time embedding and detection from the streaming-based audio forensic marking system that has been implemented on the server/client is possible.

• Journal of Broadcast Engineering
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• v.17 no.4
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• pp.659-669
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• 2012

This paper proposes the method to extract a binary audio fingerprint by combining several base binary fingerprints. Based on majority voting of base fingerprints, which are designed by mimicking the fingerprint used in Philips fingerprinting system, the proposed fingerprint is determined. In the matching part, the base fingerprints are extracted from the query, and distance is computed using the sum of them. In the experiments, the proposed fingerprint outperforms the base binary fingerprints. The method can be used for enhancing the existing binary fingerprint or for designing a new fingerprint.