• Journal of information and communication convergence engineering
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• v.5 no.3
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• pp.245-248
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• 2007

In this paper, we research the content-based analysis and classification according to the composition of the feature parameter data base for the audio data to implement the audio data index and searching system. Audio data is classified to the primitive various auditory types. We described the analysis and feature extraction method for the feature parameters available to the audio data classification. And we compose the feature parameters data base in the index group unit, then compare and analyze the audio data centering the including level around and index criterion into the audio categories. Based on this result, we compose feature vectors of audio data according to the classification categories, and simulate to classify using discrimination function.

• Proceedings of the IEEK Conference
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• 2000.06c
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• pp.125-128
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• 2000

In this paper, we proposed the high quality digital audio watermarking algorithm in the frequency domain. The spread spectrum technique is used to encrypted a stream of information by spreading the data as much of the frequency spectrum as possible. It's technique adapt well to data hiding in audio signal. We have used the perceptual model and MDCT/IMDCT for the high qualify digital audio watermarking. The proposed watermark algorithm preserved high quality audio data from watermark signal.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.82-88
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• 2002

In this paper, we proposed a audio indexing method that is used wavelet transform for audio data retrieval. It is difficult for audio data to make a efficient audio data index because of its own particular properties, such as requirement of large storage, real time to transfer and wide bandwidth. An audio data in del using wavelet transform make it possible to index and retrieval by using the particular wavelet transform properties. Our proposed indexing method doesn't separate data to several blocks. Therefore we use both high-pass and low-pass parts of last level coefficient of wavelet transform. Audio data indexing is made by applying the string matching algorithm to high-pass part and zero-crossing histogram to low-pass part. These are transformed to the continued strings, Through this method, we described a retrieval efficiency. The retrieval method is done by comparing the database index string to the query string and then data of minimum values is chosen to the result. Our simulation decided proper comparative coefficient and made known changing of retrieval efficiency versus audio data length. The results show that the proposed method improves retrieval efficiency compared to conventional method.

• Speech Sciences
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• v.7 no.1
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• pp.5-15
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• 2000

In this paper a detailed description of collecting Korean audio-video speech data is presented. The main aim of this experiment is to collect some audio-video materials which can be used for later experiments to estimate and model the actions of the visible human articulatory organs such as mouth, lips and jaw. We collect audio-video data from seven directions separately. Twelve markers are used to trace the movements.

• IEMEK Journal of Embedded Systems and Applications
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• v.2 no.1
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• pp.31-42
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• 2007

A transform coding scheme for the transmission of audio data in ZigBee based wireless personal area networks (WPAN) is presented in this paper. Here, wavelet transform is exploited to encode the features of audio data included mainly in the low frequency region. As a result, it is confirmed that the presented scheme recovers the original audio signals much accurately while it transmits the binary data compressed as 37.5% of the entire data generated without coding scheme. Especially, the mean-squared error between the recovered and original audio data approaches $10^{-4}$ when the signal-to-noise power ratio is sufficiently high. Hence, the presented coding scheme which exploits the wavelet transform is possibly applied for high-quality audio data transmission services in a small-scale sensor network based on ZigBee. Such a result is considered to be applicable as a basic material to update the technical specifications and develop the applications of ZigBee in WPANs.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.454-457
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• 2002

In this paper, we proposed digital audio watermarking algorithm for high quality audio. Nowadays, digital watermark used to confirm to digital copyright protection, not only digital image but also digital audio is active in the digital watermarking study. In this paper, we proposed digital audio watermarking algorithm using psychoacoustics model and MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform) for the high quality audio watermark. In the proposed scheme, we used to 441KHz, 128kbps and stereo audio data for audio watermarking algorithm. Audio data is passed by MDCT; watermark can be inserted into the frequency domain with 256,1024 and 2048 interval.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.3
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• pp.53-61
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• 2002

In this paper, the authors proposed the digital audio watermarking algorithm for a high quality audio. Today, the digital watermark is used to confirm to the digital copyright protection, not only the digital image but the digital audio study is an activeness in the digital watermarking area. Especially, the watermark insertion in the digital audio area affects deeply not only a robustness but the audio quality of the watermarked audio data. Generally, the audio watermark is inserted in the frequence domain after FFT, the quality of audio data is affected by the watermark insertion. Thus, a high quality audio to be maintained at the same time, the study related a inserting of the robustness watermark happened to a hot issue. In this paper, the authors proposed the digital audio watermarking algorithm using psychoacoustic model and MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform). In the proposed scheme, the authors experimented the stereo audio file with 44.1KHz, and 128kbps for the audio watermarking algorithm proposed. When the audio data is processed by MDCT, the watermark is able to insert into the frequence domain with 256, 1024 and 2048 interval. In case of 50㎳ RMS window, it was confirmed that the difference between the original audio data and the watermarked audio data of RMS power is 0.8㏈.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.2
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• pp.461-468
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• 2000

Amounts of audio data used in several computer application have necessitated the development of audio database systems with newer features such as content-based queries and similarity searches to manage and use such data. Fast and accurate retrievals for content-based queries are crucial for such systems to be useful. Efficient content-based indexing and similarity searching schemes are keys to providing fast and relevant data retrievals. This paper present a method for indexing of Korean Traditional Music audio data based on wavelets. Also this paper present possibility of wavelet based audio indexing.

• ETRI Journal
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• v.31 no.4
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• pp.365-375
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• 2009

In this paper, a terrestrial digital multimedia broadcasting (T-DMB) multichannel audio broadcasting system based on spatial audio coding is presented. The proposed system provides realistic multichannel audio service via T-DMB with a small increase of data rate as well as backward compatibility with the conventional stereo-based T-DMB player. To reduce the data rate for additional multichannel audio signals, we compress the multichannel audio signals using the sound source location cue coding algorithm, which is an efficient parametric multichannel audio compression technique. For compatibility, we use the dependent property of an elementary stream descriptor, and this property should be ignored in a conventional T-DMB player. To verify the feasibility of the proposed system, we implement the T-DMB multichannel audio encoder and a prototype player. We perform a compatibility test using the T-DMB multichannel audio encoder and conventional T-DMB players. The test demonstrates that the proposed system is compatible with a conventional T-DMB player and that it can provide a promisingly rich audio service.