• 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.

• Journal of the Korea Society of Computer and Information
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• v.22 no.10
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• pp.63-71
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• 2017

In this paper, we develop the system for audio volume equalization of video on demand(VoD) content and propose the solution for it. In recent years, there has been a steady increase in the number of VoD users in addition to linear channels. However, viewers ought to sit in an uncomfortable way, adjusting the volume intermittently while they are broadcasted. Sudden changes of volume occur between the broadcasting channels, the programs from the co-channel, or the linear channels and the VoDs. Especially, upsurged dissatisfaction from the televiewers has been found due to the unequalized volume when shifting between the linear channel and the VoD. In order to solve this problem, multilateral efforts were put forth, such as a system for keeping the volume at a certain level in digital broadcasting program has been legislated domestically. It leads success in equalizing linear channel volume. On contrary, too little notice has been taken for distorted volume problem of video on demand(VoD) content. In this paper, we developed and applied the volume equalization system into VoD content to achieve uniformization, a similar condition with linear channel(-24LKFS). This suggestion helped uneven current of volume which was in the stage -16 ~ -20LKFS to stable condition by lowering into the stage of -24LKFS. It also brought 20% increase in perspective of volume quality satisfaction level.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.2
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• pp.437-444
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• 2010

The audio data have been communicated using analog methods or simple protocols. However, with the advent and improvement of various multimedia functions, many audio devices have been integrated into a mobile handset in which interconnection lines are very complicated. Conventional point-to-point connections such as $I^2S$ and PCM demand more power consumption whenever more devices are attached. In this paper, we design a common bus digital audio interface that communicates with only two wires and employs the clock gear method to reduce bus power consumption. The comparison results show that the proposed common bus connection can reduce more than 30% of power consumption as compared with point-to-point connection if more than three devices are connected.

• Proceedings of the IEEK Conference
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• summer
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• pp.199-203
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• 2004

In this paper, we only describe the digital block of two-channel 18-bit analog-to-digital (A/D) converter employing sigma-delta method and xl28 decimation. The device contains two fourth comb filters with 1-bit input from sigma­delta modulator. each followed by a digital half band FIR(Finite Impulse Response) filters. The external analog sigma-delta modulators are sampled at 6.144MHz and the digital words are output at 48kHz. The fourth-order comb filter has designed 3 types of ways for optimal power consumption and signal-to-noise ratio. The following 3 digital filters are designed with 12tap, 22tap and 116tap to meet the specification. These filters eliminate images of the base band audio signal that exist at multiples of the input sample rate. We also designed these filters with 8bit and 16bit filter coefficient to analysis signal-to-noise ratio and hardware complexity. It also included digital output interface block for I2S serial data protocol, test circuit and internal input vector generator. It is fabricated with 0.35um HYNIX standard CMOS cell library with 3.3V supply voltage and the chip size is 2000um by 2000um. The function and the performance have been verified using Verilog XL logic simulator and Matlab tool.

• Journal of KIISE:Information Networking
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• v.33 no.2
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• pp.139-148
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• 2006

Recently with the advancement of local wireless communication technologies, digital trends of audio contents and devices are paving the way for new experiences to network-based audio streaming services. In this paper, we present the Wireless Audio Sharing (WASH) technology over Bluetooth and Wireless Local Area Network (WLAN). WASH system provides an audio sharing mechanism over multiple users and also provides stereo audio streaming between Bluetooth-enabled devices and Universal Plug and Play (UPnP) media devices connected to wired/wireless LANs. To achieve these, we extended the Bluetooth audio distribution profile and combined the extended profile with the UPnP AV architecture. With the implementation of this WASH technology, we show some experimental results of the stereo audio streaming in a real environment.

• Asia-pacific Journal of Multimedia Services Convergent with Art, Humanities, and Sociology
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• v.8 no.4
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• pp.151-160
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• 2018

The development of digital technology and the Internet has brought about a great change in the content industry. In order to keep pace with these changes, the copyright law has undergone several revisions, and the concept of "digital sound transmission" was introduced in the 2006 revision. However, in the current law, digital audio transmission is problematic in that the criteria for distinguishing between broadcasting and transmission is abstract and unclear. This ambiguity makes it difficult to judge the legal status of new music webcasting service such as "Free Litsen". Although these services are positioned as digital audion transmission, they have created a new concept of pseudo transmission because of its similarity to the audio transmission in its convenience and utility. These problems stem from the imbalance of between the development of technology and the legal system, so the change of the legal system is inevitable. Thus, this study discusses US copyright law and related cases, and then suggests solutions for pseudo transmission problems. This study suggests legislative criteria for pseudo transmission problems and legislative measures that can reduce the actual damage to the music market.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.271-278
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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 sign processing approach. From the 20 seconds query audio file, the audio signal is segmented into 23ms frame with non-overlapped hamming window and 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 can verify the superior performance of the proposed method that provides near 90% success rate for the genre classification which means 10%∼20% improvements over the previous methods. For the case of actual user system environment, feature vector is extracted from the random interval of the query audio and it shows overall 80% success rate except extreme cases of beginning and ending portion of the query audio file.

• ETRI Journal
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• v.28 no.3
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• pp.364-366
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• 2006

This letter introduces an efficient transmission of traffic information through a terrestrial digital multimedia broadcasting system, which is a multimedia and mobility empowered option of the European digital audio broadcasting system. By adapting Korean characteristic traffic information into the transport protocol expert group messages in the traffic information delivery, a highly efficient traffic information system was implemented and tested in Korea.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.54-60
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• 2004

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio, current digital audio is always restricted by sampling rate and bandwidth. This restriction normally results in low sample rate audio or calls for the data compression scheme such as MP3. However, they can only reproduce a lower frequency range than a regular CD quality because of the Nyquist sampling theory. Consequently they lose rich spatial information embedded in high frequency. The propose of this paper is to propose efficient high frequency enhancement of low sample rate audio using n adaptive filtering and DWT analysis and synthesis. The proposed algorithm uses the LMS adaptive algorithm to estimate the missing high frequency contents in DWT domain and it then reconstructs the spectrally enhanced audio by using the DWT synthesis procedure. Several experiments with real speech and audio are performed and compared with other algorithm. From the experimental results of spectrogram and sonic test, we confirm that the proposed algorithm outperforms the other algorithm and reasonably works well for the most of audio cases.