• Proceedings of the KIEE Conference
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• 1989.07a
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• pp.486-490
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• 1989

In this paper, a program controlling an auto-audio media - cassette deck - by a 16 bit personal computer is studied in order to execute audio and visual learning in CAI. The results of this study are as follows. 1. Audio and visual learning is executed efficiently in CAI. 2. Access rate of voice information to text/image information is about 98% and 60% in "play" and "fast forward" respectively. 3. In "fast forward", quality of a cassette tape affects voice information access rate in propotion to motor driving speed. 4. Synchronizing signal may be mistaken by defects of tape itself.

• 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.

• Proceedings of the KSLP Conference
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• 1995.11a
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• pp.141-153
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• 1995

CSpeech is a software package that implements an audio waveform/speech analysis workstation on an IBM Personal Computer or hardware compatible computer. Features include digitizing audio waveforms on single or multiple channels, displaying the digitized waveforms, playing back audio waveforms from selected intervals of sing1e channels, saving and retrieving waveforms from binary format disk files, and analysing audio waveforms for their temporal and spectral properties. The distinguishing characteristics of CSpeech are its support for multiple channels, minimal restrictions on sample rate and waveform duration support fur a variety of hardware configurations, fast graphics display, and its user- extensible menu- based command structure.

• 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.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.329-332
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• 2005

In this paper, we describe implementation of personal media gateway (PMG) for applications over BcN networks. PMG is a TV based set-top terminal, which enables transmission of Full D1 high quality video and audio at the speed of maximum 2Mbps. It supports SIP protocol and QoS for the BcN networks. The hardware of the PMG consists of host module, audio/video codec processing module, DTMF module, and remote control I/O module. H.263 and MPEG4 software are implemented in DSP as codec for hi-directional communication and streaming, respectively. G.711 and Ogg-Vorbis are implemented as audio codec. We examined the quality of video using the Video Quality Test Equpment, which was developed by KT Convergence Lab. The experimental results show the video quality of MOS 4.1 and audio quality of MOS 4.3. We expect that PMG will be prospective business models, and create new customer value.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.237-244
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• 2002

In this paper we mention the optimization of MPEG-4 VM (Moving Picture Expert Group-4 Verification Model) GA (General Audio) AAC (Advanced Audio Coding) encoder and the design of the decoder for PDA (Personal Digital Assistant) using MPEG-4 VM source. We profiled the VMC source and several optimization methods have applied to those selected functions from the profiling. Intel Pentium III 600 MHz PC, which uses windows 98 as OS, takes about 20 times of encoding time compared to input sample running time, with additional options, and about 10 times without any option. Decoding time on PDA was over 35 seconds for the 17 seconds input sample. After optimization, the encoding time has reduced to 50% and the real time decoding has achieved on PDA.

• 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 Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.638-643
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• 2007

With development of high display quality of TV and Monitor, personal audio system is arising great interest. In this study, we applied a method to make a good bright zone around the user and dark zone to other region by maximizing the ratio of sound energy between the bright and dark zone. We have attempted to use a line speaker array system to localize the sound in our listening zone. It depends on the size of the zone and array parameters, for example, array size, speaker spacing, wave length of sound.

• 한국정보통신설비학회:학술대회논문집
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• 2006.08a
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• pp.141-144
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• 2006

The personal digital recorder is a consumer electronics device that records television shows to a hard disk in digital format. In this paper, we propose an implementation method of personal digital recorder for HDTV. The proposed personal recorder includes CPU and system control modules, graphics and display module, audio DSP module, digital I/O module, NIM module, graphic software library, and embedded software modules for providing a lot of personal digital recorder functions such as live or reserved recordings, browsing of recorder content list, trick lay and time shifting. Especially, combining trick play with time shifting makes much more convenient functions such as pausing live TV, instant replay of interesting scenes, and skipping advertising.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.