• Journal of Korea Multimedia Society
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• v.4 no.4
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• pp.297-305
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• 2001

Recently advances in high-speed networks and multimedia computer technologies allow new types of multimedia applications to manipulate large volumes of multimedia data. However, in the real time and/or the heterogeneous data transmissions, there are many difficulties such as network transmission delay, the implementation difficulties, and so on. To solve these problems, in this paper, we extend the method of the multimedia service design which is proposed by OMG. To do this, we suggest an efficient real time audio/video stream framework, called Smart Explorer, based un CORBA and JMF Java Media API. And we separate the transmission path of control data from that of media data and use RTP/RTCP protocol for efficient real time audio/video transmission. Also we show the appropriate implementation of the audio/video stream system based on our suggested framework Smart Explorer. In the future, we expect our audio/video stream system to be applied to the real time communication software such as broadcasting, distance learning, and video conferencing.

• Journal of Korea Society of Industrial Information Systems
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• v.13 no.5
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• pp.117-123
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• 2008

In this paper, we develop MPEG-4 AAC streaming player on embedded Linux processor such as mobile terminals. Moreover we show the experimental results that the player preforms the decoding processes of MPEG-4 AAC data effectively. MPEG-4 AAC technology supports a wide range encoding rates and high sound quality so it is appropriate to adopt various applications. In particular, the need in the development of the application of audio data increases according to significantly increase in devices used in mobile environments, such as cell phones and PDAs. In this environment, it is necessary to optimize the decoding processes to the ability of the terminal hardware in order to play audio data without delays. We also implement the decoding module to optimize the processor capabilities and make the player to decode and play streaming audio data from streaming server.

• Proceedings of the IEEK Conference
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• summer
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• pp.114-117
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• 2004

This paper proposes a novel algorithm for hybrid lossless audio coding, which employs integer discrete cosine transform. The proposed algorithm divides the input signal into frames of a proper length, decorrelates the framed data using the integer DCT 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 data.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1045-1048
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• 1999

In this paper, we report on a set of new algorithms to realize a nonlinear compressed-domain video/audio editor that overcomes various realization problems. For efficiency, the underlying algorithm, which uses a central data structure in the form of doubled linked lists, performs soft edits of cut and paste (which, in turn, involves soft implementations of frame type conversion) and addresses problems relating to video/audio synchronization and random access, and decoder buffer control.

• Journal of the Korean Institute of Telematics and Electronics
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• v.12 no.3
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• pp.1-7
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• 1975

In this paper, a method of writing and reproducing high density data with ordinary Audio cassette deck is discribed. In writing, the data N or NRZ code are modulated into PM code to take the positive code N(1) and the negative code N(0) are taken from the complement of the NRZ code, each of which are written into 2 channel track. In reading, the error corrected and the clock pulse can be generated from the reading pulse itself. Also, without modifying the interior circuit of the deck it is possible to use the deck in both the data and audio by adapting a few simple circuits. Over the range of 25HZ-4KHZ, it was possible to write and reproduce at the speed of 787 bps transmission rate.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.202-209
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• 2005

The rapid development of digital media and communication network urgently brings about the need of data certification technology to protect IPR (Intellectual property right). This paper proposed a new watermarking method for embedding owner's audio signal. Because this method uses an audio signal as a watermark to be embedded, it is very useful to claim the ownership aurally. And it has the advantage of restoring audio signal modified and especially removed by image removing attacks by applying our LBX(Linear Bit-expansion) interleaving. Three basic stages of our watermarking include: 1) Encode . analogue owner's audio signal by PCM and create new digital audio watermark, 2) Interleave an audio watermark by our LBX; and 3) Embed the interleaved audio watermark in the low frequency band on DTn (Discrete Haar Wavelet Transform) of image. The experimental results prove that this method is resistant to lossy JPEG compression as standard image compression and especially to cropping and rotation which remove a part of Image.

• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.522-525
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• 2009

This paper proposed the content-based method of classifying filthy Phornographic video, which causes a big problem of modern society as the reverse function of internet. Audio data was used to extract the features from Phornographic video. There are frequency spectrum, autocorrelation, and MFCC as the feature of audio used in this paper. The sound that could be filthy contents was extracted, and the Phornographic was classified by measuring how much percentage of relevant sound was corresponding with the whole audio of video. For the experiment on the proposed method, The efficiency of classifying Phornographic was measured on each feature, and the measured result and comparison with using multi features were performed. I can obtain the better result than when only one feature of audio was extracted, and used.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.3-12
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• 2001

Over the last few years, digital audio watermarking has become an interesting issue In many application areas, including digital broadcasting. This is primarily motivated by a need to provide copyright protection of digital audio content. Digital watermarking is a technique to embed copyright or other information int? the underlying data. Several possible audio watermarking techniques have been developed including spread spectrum watermarking, echo watermarking, phase coding, and patchwork. In this paper, we describe some requirements of digital audio watermarking as a tool for copyright protection of broadcasting content, and compare popular audio watermarking algorithms in some significant aspects.

• Journal of the Korean Society for information Management
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• v.26 no.3
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• pp.169-188
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• 2009

The study proposed the algorithm for automatically summarizing the audio information from a video and then conducted an experiment for the evaluation of the audio extraction that was constructed based on the proposed algorithm. The research results showed that first, the recall and precision rates of the proposed method for audio summarization were higher than those of the mechanical method by which audio extraction was constructed based on the sentence location. Second, the proposed method outperformed the mechanical method in summary making tasks, although in the gist recognition task(multiple choice), there is no statistically difference between the proposed and mechanical methods. In addition, the study conducted the participants' satisfaction survey regarding the use of audio extraction for video browsing and also discussed the practical implications of the proposed method in Internet and digital library environments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.6
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• pp.1391-1396
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• 2010

Recently, communication components prefer Bluetooth or Zigbee for PAN. However, using these makes expensive and complicated products such as audio equipments, mobile phones, PC, etc. for transmitting simple messages with low rate. In this paper, we propose wireless communication method using ultrasonic in the audio frequency limit band with speakers and microphones which are in products. We suggest transmitting and receiving methods in the audio frequency limit band for transmitting data without affecting audio signal, and implement an ultrasonic communication modem. Finally, we verify the performance of the ultrasonic communication modem by experiments in an environment with background noise.