• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.619-621
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• 2013

This paper is proposing a real-time audio processing system for TCP/IP with server-client. The server sends the audio data packet which is the same size each time while playing the audio data. And the client plays the received audio data from the server. In general, The receiving speed of audio data packet is faster than processing the audio data. So, the unstable playback is occurred when playing the received audio data at the moment. In order to overcome this problem, the double buffering method is proposed.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.605-607
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• 2004

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era made of home theater system and the digital audio broadcasting (DAB). It is impossible to analyze the stability of the digital audio amplifier, which is based on the PWM signal processing. To solve this problem, the digital audio amplifier is analyzed using variable structure control theory which is one of nonlinear system theories. Moreover, the magnitude and the frequency of ripple signal, which generated by hysteresis in the comparator, is obtained using describing function which is useful to represent the input-output relation of nonlinear system.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.5 no.1
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• pp.37-42
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• 2006

A reversible audio watermarking algorithm is presented in this paper. This algorithm transforms the audio signal with the integer wavelet transform first in order to enhance the correlation between neighbor audio samples. Audio signal has low correlation between neighbor samples, which makes it difficult to apply difference expansion scheme. Second, a novel difference expansion scheme is used to embed more data by reducing the size of location map. Therefore, the difference expansion scheme used in this paper theoretically secures high embedding capacity under low perceptual distortion. Experiments show that this scheme can hide large number of information bits and keeps high perceptual quality.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.4
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• pp.385-391
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• 2007

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era. Since the digital audio amplifier is based on the PWM signal processing, it is improper to analyze the principle of signal generation using linear system theories. In this paper, a class D digital audio amplifier based ADSM (Advanced Delta-Sigma Modulation) is considered. We first model the digital audio amplifier and then explain the operation principle using variable structure control algorithm. Moreover, the ripple signal generated by the hysteresis in the comparator has a significant effect on the system performance. Thus, we present a method to find the magnitude and the frequency of the ripple signal using describing function. Finally, simulations and experiments are provided to show the validity of the proposed methods.

• ETRI Journal
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• v.28 no.5
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• pp.664-667
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• 2006

We propose an efficient audio transcoding algorithm that can convert audio streams from terrestrial digital television broadcasting service stations to those for terrestrial digital multimedia broadcasting hand-held receivers. The proposed algorithm avoids the complicated psychoacoustic analysis by calculating the scalefactors of the bit-sliced arithmetic coding encoder directly from the signal-to-noise ratio parameters of the AC-3 decoder. The bit-allocation process is also simplified by cascading the nested distortion control loop. Through subjective evaluation, it is shown that the proposed algorithm provides comparable audio quality to tandem coding but it requires much smaller complexity.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2
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• pp.77-82
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• 1999

A sinusoidal model has been widely used in the analysis and synthesis of speech and audio signals, and becomes one of the efficient candidates for high quality low bit rate audio coders. One of the crucial steps in the analysis and synthesis using a sinusoidal model is the detection of tonal components. This paper proposes an efficient method for the analysis and synthesis of audio signals using a sinusoidal model, which uses psychoacoustic criteria such as masking effect, masking index, and JNDf(Just Noticeable Difference in Frequency). Simulation results show that the proposed method reduces the number of sinusoids significantly without degrading the quality of the synthesized audio signals.

• Journal of the Korean Society for Library and Information Science
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• v.44 no.4
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• pp.95-114
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• 2010

Multimedia searching is an important activity on the Web, especially among the younger generation. The purpose of this study aims to examine college students’ multimedia information needs and searching on the Internet. While there is a clear pattern among students with respect to their multimedia uses, searching sources, relevance criteria and searching barriers, some differences exist especially according to searching of different multimedia types such as image, audio and video. For multimedia uses, information/data-focused uses are frequently found in image and video, while the use of audio is mainly for object-focused searches. As multimedia searching sources, audio and video files present a similar pattern of being high in media specific searching sources and low in generic search engines. Browsing through related blogs and homepages is an important part of searching for media files accounting for approximately 20% of total search for each media. The relevance criteria used by study participants when search for image files was primarily concerned with topicality while the contextual and media quality in the audio and video types are also considered important. Searching barriers for audio and video files are categorized into three broad aspects, including access and search quality, preview limitations and collection limitations, while obstacles for image files searching include access difficulties and low qualities of various collection.

• Journal of Korean Society for Quality Management
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• v.44 no.4
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• pp.951-963
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• 2016

Purpose: This study was conducted to analyze the potential customer's satisfaction for the concepts of smart audio features and utilize the results when developing the customer-oriented products. Methods: 16 different features were derived via the market research and professionals' interviews. The most satisfactory features were selected through "Kano model", the relative importance of Customer Satisfaction Coefficient, and respondents' preferences from 339 valid survey answers. Results: 15 out of the 16 features were categorized as attractive attribute. "'User Recognizing' and 'Strengthen Linking' groups", such as Auto connection with Smart-phone music player, Synchronization of TV & Audio, and Volume control situational awareness, were shown to provide higher satisfactions to those potential customers. On the other hand, Group 'Integrating Function', such as Aromatherapy and Auto lighting reaction, was shown to be relatively least preferred features. Conclusion: This study enabled which features could lead to the customer satisfaction. Nevertheless, it still requires extensive analyses in different countries and diverse cultures to target the global market. The audio product planners and R&D professionals are expected to learn useful information from such studies.

• 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.