• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2012.05a
• /
• pp.314-317
• /
• 2012

최근 차량의 멀티미디어 장치들이 증가하면서 이 장치들을 광 네트워크로 연결하여 멀티미디어 데이터를 송 수신해서 사용할 수 있는 MOST(Media Oriented Systems Transport) 네트워크를 적용한 차량들이 늘어나고 있다. MOST 네트워크는 최근 자동차 멀티미디어 시스템에 넓게 사용되고 있는 통신 시스템으로서, 동기 및 비동기 데이터를 동시에 전송할 수 있고, 최근에는 150Mbps를 전송할 수 있는 MOST150 네트워크를 이용한 연구가 활발히 진행되고 있다. 본 논문에서는 MOST150 네트워크에서 Audio 데이터 입출력을 제어하기 위한 알고리즘을 설계 및 구현하였다. Audio 데이터를 제어하는 방식은 ADC(Analog to Digital Converter)를 통해 Audio 데이터가 들어오면 IOC(IO Companion)를 통해 INIC으로 Audio데이터를 전달한다. INIC은 MOST150 네트워크로 데이터를 전송하고 그렇게 보내진 Audio 데이터를 MOST150 네트워크 내부의 다른 장치에서 INIC을 통해 데이터를 수신하여 DAC(Digital to Analog Converter)를 통해 Audio 장치에서 소리가 나는 것을 테스트하여 정상적으로 동작함을 확인하였다.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.17 no.2
• /
• pp.225-232
• /
• 2017

Recently, the development of digital transmission technology of audio signals and the introduction of audio network equipment using digital transmission technology have been made. As a result, audio network technology and equipment are actively applied to the design and construction of audio systems. The meeting room is a place where a large number of participants exchange opinions and communicate with each other. In addition to using an electric acoustic device such as a microphone and a speaker, it improves the intelligibility of the conference room through an example using an audio network.

• Journal of the Korean Society for information Management
• /
• v.26 no.3
• /
• pp.169-188
• /
• 2009

The study proposed the algorithm for automatically summarizing the audio information from a video and then conducted an experiment for the evaluation of the audio extraction that was constructed based on the proposed algorithm. The research results showed that first, the recall and precision rates of the proposed method for audio summarization were higher than those of the mechanical method by which audio extraction was constructed based on the sentence location. Second, the proposed method outperformed the mechanical method in summary making tasks, although in the gist recognition task(multiple choice), there is no statistically difference between the proposed and mechanical methods. In addition, the study conducted the participants' satisfaction survey regarding the use of audio extraction for video browsing and also discussed the practical implications of the proposed method in Internet and digital library environments.

• Proceedings of the Korea Society of Design Studies Conference
• /
• 2005.05a
• /
• pp.212-213
• /
• 2005

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 1999.11a
• /
• pp.199-203
• /
• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• Proceedings of the KIEE Conference
• /
• 2005.10b
• /
• pp.358-360
• /
• 2005

The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, uses 1 bit representation with a sampling frequency of 2.8224MHz (64${\times}$44.1kHz). For multi-rate PCM (Pulse Code Modulation) input such as 8${\sim}$192kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital mter structure composed of sample-rate converter and interpolaton filter for the DSD encoder with multi-rate (8${\sim}$192kHz) PCM input, without a external sample-rate converter.

• Journal of Broadcast Engineering
• /
• v.8 no.3
• /
• pp.217-227
• /
• 2003

This paper describes an implementation of the slide-show functionality, which is one of the services that can be provided by the Digital Multimedia Broadcasting (DMB). While the existing analog radio broadcasting services provide audio only, DMB slide-show is the functionality that can deliver still images associated with the audio. For example, it can deliver the photographs of the singer, album cover images, or the lyrics of the song that correspond to the audio. There are two modes for the transmission of the slide-show. Firstly. the program-associated data (PAD) field within the DMB audio frame can be utilized and secondly, the slide-show data can be transmitted, after being multiplexed, with other service data as individual data stream separated from the audio. This paper describes PC-based implementations of a transmitter-side module that inserts slide-show data into the PAD area within audio bitstream and a receiver-side application module that plays the slide-show through decoding the PAD within the received audio bitstream and demonstrates their validity through experiments.

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

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2017.05a
• /
• pp.656-659
• /
• 2017

In this paper, we propose a digital method for transmitting audio signals using LED visible light communication system. In the proposed method, we compare the method for transmitting audio signal in analog signal and the method for transmitting by digital signal. When amplifying the audio sound and transmitting the analog signal using the LED visible light communication, attenuation corresponding to the transmission distance occurs, and there is a disadvantage that it is noisy. In order to overcome this, we propose a method for transmitting digital audio signals. The proposed method has the advantage of reducing the influence of noise, but it turned out that it is affected much by the LED blinking speed. Various methods to overcome this need to be continuously studied.

• Phonetics and Speech Sciences
• /
• v.6 no.2
• /
• pp.21-28
• /
• 2014

We propose a novel scheme for digital audio authentication of given audio files which are edited by inserting small audio segments from different environmental sources. The purpose of this research is to detect inserted sections from given audio files. We expect that the proposed method will assist human investigators by notifying suspected audio section which considered to be recorded or transmitted on different environments. GMM-UBM and GSV-SVM are applied for modeling the dominant environment of a given audio file. Four kinds of likelihood ratio based scores and SVM score are used to measure the likelihood for a dominant environment model. We also use an ensemble score which is a combination of the aforementioned five kinds of scores. In the experimental results, the proposed method shows the lowest average equal error rate when we use the ensemble score. Even when dominant environments were unknown, the proposed method gives a similar accuracy.