• The Journal of the Acoustical Society of Korea
• /
• v.15 no.2
• /
• pp.55-60
• /
• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1994.06a
• /
• pp.912-918
• /
• 1994

Poor speech intelligibility in an air traffic control room is frequently a result of many, quite different causes and occasionally leads to complaints of the controller personnel. The paper describes a sequence of successful tasks performed in a local control room. The initial measurements included an investigation of the background noise (caused by fans, air condition, computer and radar equipment) and performance checks of the electronic audio and communication equipment with respect to the audio transmission behavior. The spectral composition of the noise as well as the characteristics of the audio communication path between the controllers and the pilots(which showed a loss of spectral information in the audio band due to built-in notch filters for the suppression of control tones) required adaptations of the amplitude behavior of the amplifiers through user adjustable tone controls. The radar console fans, which contributed significantly to the overall noise floor of the room, underwent a substantial reconstruction by replacing the tight mounting with an elastic double suspension, reducing the noise level by 50%. Finally, a possible source of untimely fatigue of the controllers during their working hours has been found in strong spectral components of the noise above the audio band, radiated by numerous video monitors in the control through vibrating components excited by the line frequency of the video signal.

• Proceedings of the IEEK Conference
• /
• 2002.07c
• /
• pp.1567-1570
• /
• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

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

• Journal of the Korea Society of Computer and Information
• /
• v.19 no.10
• /
• pp.159-168
• /
• 2014

The Web is a way to share documents and communicate. However, voice and video data can be transmission in real time and currently being developed by the objects and the objects that interact to further develop the Internet. Existing video and audio programs to transmission data to the interface of different types of systems a lot of constraint condition on the cost of the interface, extensibility. In this paper, voice and audio transmission system a different operating system, improve the constraints of the ERP system compatibility and extensibility is a open source based system developed by the research. The program is different types of systems and interface, extensibility with program design and development methodologies, and open source-based system composed This system is good for cost saving and extensibility. Therefore, systems research and development, Extensibility and excellent on the interface, system design and development methodologies, such as real-time video conferencing, HMI, and take advantage of your video available from SNS.

• Journal of Broadcast Engineering
• /
• v.8 no.1
• /
• pp.72-79
• /
• 2003

The Eureka-147 Digital Audio Broadcasting(DAB) system is the new international standard for mobile broadcasting services including high-quality audio, program associated data and other multimedia data. In this paper, we design the transport frame structures for the mobile multimedia services by using the configuration of the Eureka-147 DAB multiplex and MPEG system specifications, and then compare their performances in terms of functionality and overhead. Especially, we suggest and analyze the effective transport structures, which are improved in the efficiency of media multiplexing architecture, by removing the functionally overlapped parts between DAB and MPEG systems. In addition, we evaluate the transmission environments of various DAB data channels and demonstrate the transmission error effects on the low bit rate video stream.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.7
• /
• pp.592-602
• /
• 2003

This paper aims to describe the basic structure of novel object-based 3D audio broadcasting system To overcome current uni-directional audio broadcasting services, the object-based 3D audio broadcasting system is designed for providing the ability to interact with important audio objects as well as realistic 3D effects based on the MPEG-4 standard. The system is composed of 6 sub-modules. The audio input module collects the background sound object, which is recored by 3D microphone, and audio objects, which are recorded by monaural microphone or extracted through source separation method. The sound scene authoring module edits the 3D information of audio objects such as acoustical characteristics, location, directivity and etc. It also defines the final sound scene with a 3D background sound, which is intended to be delievered to a receiving terminal by producer. The encoder module encodes scene descriptors and audio objects for effective transmission. The decoder module extracts scene descriptors and audio objects from decoding received bistreams. The sound scene composition module reconstructs the 3D sound scene with scene descriptors and audio objects. The 3D sound renderer module maximizes the 3D sound effects through adapting the final sound to the listner's acoustical environments. It also receives the user's controls on audio objects and sends them to the scene composition module for changing the sound scene.

• Proceedings of the IEEK Conference
• /
• 2008.06a
• /
• pp.633-634
• /
• 2008

In this paper, we describe implementation of a network Speaker for easily read streaming audio data from the network. The Network Speaker uses MAXIM company's DS80C400 for network control and MAX542 for audio data play. The DS80C400 network microcontroller offers TCP IPv4/6 network stack with the TINI-OS provided in ROM. The TINI-OS is adopted as an embedded operating system. Application programs are implemented by using JAVA language.

• 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.14 no.3
• /
• pp.332-341
• /
• 2009

Terrestrial digital multimedia broadcasting(T-DMB) is one of mobile broadcasting services, and the commercial service was started in December 2005 in Korea. The performance targets of T-DMB are providing VCD(video CD) quality video and FM radio quality audio. In recent years, the researches for providing high quality video or audio service on T-DMB environments have been being carried out. To provide high-quality video or audio service, some additional data should be transmitted to the receiver as well as T-DMB video and audio data. Since the data rate for one T-DMB program is low, it is important to transmit the additional data at a low bit rate. In this paper, we propose a packetizing method for efficient transmission of the additional data to provide multichannel audio service on T-DMB environment.