• Journal of the Korea Society of Computer and Information
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• v.13 no.7
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• pp.91-98
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• 2008

In digital broadcasting, it is possible to multiplex maximum one hundred audio or music programs into MPEG-2 transport stream, which is suitable for transmitting through one channel. In order to check if multiplex music programs are transmitted well, we need a multiplex audio monitoring system that monitors the programs in real-time. In analog broadcasting, we have used hardware-based audio monitoring system for a small number music programs. However, the effectiveness of hardware-based audio monitoring system from the cost and function viewpoint is so low that a new system is needed for digital broadcasting. In this paper, we have designed and implemented a software-based audio monitoring system to satisfy these requirements. In this implementation, only one PC is used without other hardware facilities, and the system monitors digital broadcasting music programs effectively. Transmitted digital broadcasting streams are demultiplexed into many music programs and the realtime value of audio level and packet error information for these programs are displayed in the screen. Thus, the system detects and shows the abnormal transmitting programs automatically. Simulation results show that effective realtime multiplex audio monitoring is possible for digital broadcasting music programs.

• International Journal of Control, Automation, and Systems
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• v.5 no.1
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• pp.61-69
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• 2007

In this paper, we developed not only a reliable sound localization system including a VAD(Voice Activity Detection) component using three microphones but also a face tracking system using a vision camera. Moreover, we proposed a way to integrate three systems in the human-robot interaction to compensate errors in the localization of a speaker and to reject unnecessary speech or noise signals entering from undesired directions effectively. For the purpose of verifying our system's performances, we installed the proposed audio-visual system in a prototype robot, called IROBAA(Intelligent ROBot for Active Audition), and demonstrated how to integrate the audio-visual system.

• Journal of the Korean Society for information Management
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• v.15 no.1
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• pp.149-164
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• 1998

The study aims to construct a system which can provide multimedia materials, specifically, digitalized video and audio materials on the internet. To accomplish this objective, it investigates technology on constructing a VOD/AOD system, current situations on video and audio data management in domestic and internatinal broadcasting institution and information centers. The study proposes a VOD/AOD system which can effectively manage and disseminate these materials on the internet.

• The Journal of the Acoustical Society of Korea
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• v.32 no.4
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• pp.346-353
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• 2013

This paper deals with the implementation of a high-quality audio system for karaoke. For improving the key/tempo changes performance, we separated the audio into many musical instrument channels. By separating musical instrument channels, high-quality key/tempo changes can be achieved and we confirmed this using the cross-correlation distribution and the MOS evaluation. The improved audio system was implemented using the TMS320C6747 DSP with fixed/floating-point operations. The implemented audio system can perform the multi-channel WMA decoding, the MP3 encoding/decoding, the wav playing, the EQ, and the key/tempo changes in real time. The WMA channels used for processing the separated instrument channels. The audio system includs the MP3 encoding/decoding function for playing and recording and the wav channel for the effect sound.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.165-172
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• 2016

This paper presents an implementation of a live audio streaming system using the Raspberry Pi 3 embedded computer. This system is a live streaming system not file-based streaming. This is a push streaming system which converts the incoming analog audio signal to digital samples and broadcasts them to multiple connected users concurrently. Since the server software is developed in Java language, it can be installed on any other embedded computers without any modification. We concluded that ESCatcher can service live streaming about 60 users concurrently through calculations and experiments, And also we achieved the delay time of a little bit more than 40ms between arrival of audio source and play on the android device.

• Journal of the Korea Society of Computer and Information
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• v.10 no.5 s.37
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• pp.353-362
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• 2005

Recently, study of Bimodal recognition has become very active. In this paper we propose a Bimodal user recognition system that uses face information and audio information. Face recognition consists of face detection step and face recognition step. Face detection uses AdaBoost to find face candidate area. After finding face candidates, PCA feature extraction is applied to decrease the dimension of feature vector. And then, SVM classifiers are used to detect and recognize face. Audio recognition uses MFCC for audio feature extraction and HMM is used for audio recognition. Experimental results show that the Bimodal recognition can improve the user recognition rate much more than audio only recognition, especially in the Presence of noise.

• Journal of Aerospace System Engineering
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• v.11 no.6
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• pp.1-9
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• 2017

The ICS, Intercom is the equipment which mixes and distributes the audio signal from other LRUs and plays the Voice Messages. Henceforth, it is of immense contributory importance to the pilots. Especially, the audio routing, which controls On/Off mode of each audio channel, is significant in executing a pilots' mission. But the audio routing process is quite complicated as it has the interface combination of many control signals. Underthecondition, the exceptional handling becomes difficult, which decreases maintainability and productivity. In the present work, to prevent such a situation, the author suggests amethodology,whichwillhavealower impact when the software is changed and provides high maintainability and productivity for audio routing processing.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.70-75
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• 2007

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.8 no.1
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• pp.75-81
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• 2008

This paper presents a top-down approach to implement an MPEG-2 audio encoder in VLSI. As the algorithm of an MPEG-2 audio encoder is heavy-weighted and heterogeneous(to be mixture of several strategies), the encoder design process is undertaken carefully from the algorithmic level to the architectural level. Firstly, the encoding algorithm is analyzed and divided into sub-algorithms, called tasks, and the tasks are partitioned in the way of reusing the same designs. Secondly, the partitioned tasks are scheduled and synthesized to make the most efficient use of time and space. In the end, a real-time 5 channel MPEG-2 audio encoder is designed which is a heterogeneous multiprocessor system; two hardwired logic blocks and one specialized DSP processor.