• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.5
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• pp.327-329
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• 2005

Conventional CDDA(Compact Disc Digital Audio) system has limitation which means that bandwidth and resolution of the sign릴 are determined by the sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. Though, new medium such as DVD-audio is developed for high-resolution audio recording, it has high complexity and difficulty in manufacturing process. So, CDDA is a widely used medium for high fidelity audio yet. In this paper, we design a new encoding system for high-resolution audio signal which has backward compatible with conventional CDDA. By evaluating for the encoding and decoding process. we verify the availability of our proposed system.

• Journal of Broadcast Engineering
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• v.24 no.7
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• pp.1228-1236
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• 2019

Recently, media platforms such as YouTube and Twitch provide services that can generate personal revenue by utilizing media content produced by individuals. In this regard, interest in the copyright of media content is increasing. In particular, in the case of an audio source, competition for securing audio source copyright is fierce because it is an essential element for almost all media content production. In this paper, we propose a method to store the audio source and its signature using a blockchain and distributed storage system to verify the copyright of music content. To identify the possibility of extracting the audio signature of the audio source and to include it as blockchain transaction data, we implement the audio source and its signature file upload system based on the proposed scheme. In addition, we show the effectiveness of the proposed method through experiments on uploading and retrieving audio files and identify future improvements.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

• Transactions on Electrical and Electronic Materials
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• v.9 no.3
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• pp.123-127
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• 2008

The paper is to solve the problem of existing audio signal transfer system which has a difficulties of system organization and the increase of additional install cost and unfriendly interior. To solve the existing system, we drew the new audio signal transfer system based on PLC and evaluated it. A transmitter and a receiver were designed using the PLC chip INT5500CS. An audio signal transfer system was configured with a CD player to which audio signals are sent from the transmitter and a speaker connected to the receiver. For performance evaluation of this system, a USBPre external sound card and Smaart Live 5 which is a PC-based sound measuring program were added. As a result of our experiment, the measured signal level is $2{\sim}3$ dB lower than reference signal, latency is 16.69 ms, and the specific character of coherency is bad in high frequency band. Otherwise, this system transmits and receives signals over 90 % in good condition as a result of measuring pink noise, frequency (1 kHz), and phase, magnitude. In view of the result so far achieved, the system designed this study has excellent performance, it resolves defect of existing audio signal transfer system.

• ETRI Journal
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• v.27 no.5
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• pp.635-638
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• 2005

Spatial audio coding (SAC) is an extremely high compact representation of encoded multi-channel audio material. This paper suggests a multi-channel audio service in the terrestrial digital multimedia broadcasting (T-DMB) system using a novel SAC tool, which is called a virtual source location information (VSLI)-based SAC tool. Intensive experiments are presented to evaluate the validity of the proposed VSLI-based SAC tool, and prototypical systems are also presented to demonstrate the reliability of the proposed multi-channel T-DMB system in real applications.

• Proceedings of the KSPS conference
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• 2006.11a
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• pp.113-116
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• 2006

In this paper, we first implement an audio playback system for virtual reality by providing 3D audio effects to listeners. In general, such a 3D audio playback system utilizes a sound localization technique using head related transfer function (HRTF) to generate 3D audio effect. However, the 3D audio effect is degraded due to the crosstalk in the stereo loudspeaker environment. To enhance the 3D sound effect, we implement the crosstalk cancellation technique proposed by Atal and Schroeder and apply it to the 3D audio system.

• Journal of information and communication convergence engineering
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• v.18 no.2
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• pp.115-122
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• 2020

Audiovisual (AV) facilities such as TVs and signage are installed in various public places. However, audio cannot be used to prevent noise and interference from individuals, which results in a loss of concentration and understanding of AV content. To address this problem, a total technique for remotely listening to audio from audiovisual facilities with clean sound quality while maintaining video and lip-syncing through personal smart mobile devices is proposed in this paper. Through the experimental results, the proposed scheme has been verified to reduce system power consumption by 8% to 16% and provide real-time processing with a low latency of 120 ms. The system described in this paper will contribute to the activation of audio telehearing services as it is possible to provide audio remote services in various places, such as express buses, trains, wide-area and intercity buses, public waiting rooms, and various application services.

• 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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• 2000.06b
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• pp.181-184
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• 2000

In this paper, we described a design of the input buffer system for efficiently dealing with MPEG audio bitstream to demux header and side information, audio data. In order to overcome the limitations of fixed-word manipulation in bitstream demuxing, we proposed a new variable length bit retrieval system with FSM sequencer supporting MPEG audio frame format, and serial buffer demuxing audio stream, FIFO circular buffer including header and side information.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1415-1419
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• 2008

In this paper, we investigate various Audio Transmission Systems to implement Audio Transmission System based on AoIP of Internet transmission technology TCP/IP Network and we design and implement a Audio transmission system based on AoIP by adopting the most efficient one. The implemented system can be applied for various professional audio systems with large-scale audio distribution network as well as small-scale PA systems. Moreover, this can be applied in various fields which need audio transmission.