• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.12
• /
• pp.232-237
• /
• 2013

The role of the window length of time-frequency transformation is important for the audio data hiding methods utilizing phase information. In this paper, the experiments for our audio data hiding method were conducted in order to evaluate the audio quality and robustness against reverberant environment. The experimental results showed the tendency that the worse audio quality but better robustness were obtained when the lengthy window was applied. The important reason for quality degradation was pre-echo which flatters the percussive sound. The results also indicated that the wireless communication theory related to the length of time-frequency transform can be applied in the field of audio data hiding and acoustic data transmission.

• 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.

• Journal of KIISE:Computing Practices and Letters
• /
• v.5 no.4
• /
• pp.440-448
• /
• 1999

본 논문은 객체지향형 분산처리 환경 하에서 오디오나 비디오 등과 같은 실시간(real-time) 스트림(stream) 데이타를 처리하는 데 필요한 소프트웨어 기반구조를 설계하고 구현한 내용을 기술한다. 본 논문에서 제시한 DAViS(Distributed Object Framework supporting Audio/Video Streaming)는, 오디오/비디오 데이타의 처리와 관련된 여러 소프트웨어 구성요소들을 분산객체로 추상화하고, 그 객체들간의 제어정보 교환경로와 오디오/비디오 데이타 전송경로를 서로 분리하여 처리한다. 분산응용프로그램 작성자는 DAViS에서 제공하는 서비스들을 이용하여, 기존의 분산프로그래밍 환경이 제공하는 것과 동일한 수준에서 오디오/비디오 데이타에 대한 처리를 표현할 수 있다. DAViS는, 새로운 형식의 오디오/비디오 데이타를 처리하는 부분을 손쉽게 통합하고, 하부 네트워크의 전송기술이나 컴퓨터시스템 관련 기술의 진보를 신속하고 자연스럽게 수용할 수 있도록 하는 유연한 구조를 가지고 있다. Abstract This paper describes the design and implementation of software framework which supports the processing of real-time stream data like audio and video in distributed object-oriented computing environment. DAViS(Distributed Object Framework supporting Audio/Video Streaming), proposed in this paper, abstracts software components concerning the processing of audio/video data as distributed objects and separates the transmission path of data between them from that of control information. Based on DAViS, distributed applications can be written in the same abstract level as is provided by the existing distributed environment in handling audio/video data. DAViS has a flexible internal structure enough to easily incorporate new types of audio/video data and to rapidly accommodate the progress of underlying network and computer system technology with very little modifications.

• ETRI Journal
• /
• v.28 no.3
• /
• pp.364-366
• /
• 2006

This letter introduces an efficient transmission of traffic information through a terrestrial digital multimedia broadcasting system, which is a multimedia and mobility empowered option of the European digital audio broadcasting system. By adapting Korean characteristic traffic information into the transport protocol expert group messages in the traffic information delivery, a highly efficient traffic information system was implemented and tested in Korea.

• Proceedings of the IEEK Conference
• /
• 2000.07a
• /
• pp.113-116
• /
• 2000

Phase noise of oscillator gives the performance degradation significantly when a high carrier frequency and low transmission rate are used. The BER(Bit Error Rates) degradation of QPSK(Quadrature Phase Shift Keying) transmission is analyzed with the oscillator phase noise level specified in downstream physical interface of LMDS(Local Multipoint Distribution Services) which is described in DAVIC(Digital Audio Visual Council). The model used for the phase noise is a power-law model. We also investigated the effects of the various transmission rates on system performance. For the transmission rate below 0.5 Mbps, the BER performance is severely degraded and we verified that the transmission rate, 20 Mbps, is adequate for the downstream of LMDS systems.

• Proceedings of the IEEK Conference
• /
• 2004.08c
• /
• pp.792-795
• /
• 2004

This paper is an interference analysis from S-DAB(Satellite-Digital Audio Broadcasting) into terrestrial IMT-2000 systems intending to use the band 2630-2 655 MHz and that could be used to determine the impact of S­DAB on terrestrial IMT-2000 in the context of co-frequency sharing through the development of pfd masks.

• Journal of Broadcast Engineering
• /
• v.11 no.3 s.32
• /
• pp.363-378
• /
• 2006

The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality audio-visual multimedia contents to the mobile environment. The system adopts MPEG-4 standard for the main video, audio and other media format. It also adopts the MPEG-4 scene description for interactive multimedia contents. The animated and interactive contents can be actualized by BIFS(Binary Format for Scene), the binary format for scene description that refers to the spatio-temporal specifications and behaviors of the individual objects. As more interactive contents are, the scene description is also needed more high bitrate. However, the bandwidth for allocating meta data such as scene description is restrictive in mobile environment. On one hand, the DMB terminal starts demultiplexing content and decodes individual media by its own decoder. After decoding each media, rendering module presents each media stream according to the scene description. Thus the BIFS stream corresponding to the scene description should be decoded and parsed in advance of presenting media data. With these reason, the transmission delay of BIFS stream causes the delay of whole audio-visual scene presentation although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique for adapting BIFS stream into expected MPEG-2 TS bitrate without any bandwidth waste and avoiding the transmission delay of the initial scene description for interactive DMB contents.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.3 s.303
• /
• pp.31-40
• /
• 2005

In this paper, we develop an approach toward JSC(Joint Source-Channel Coding) method for MPEG-4 based FGS(Fine Granular Scalability) video coding and transmission in fixed and mobile receiving environment(Digital Audio Broadcasting, DAB). The source coder used MPEG-4 FGS video codec, the channel coder used RCPC(Rate Compatible Punctured Convolution) code and the modulation method used QPSK modulation. We have considered channel environment of AWGN and mobile receiving environment. This study determined optimum Trade-off point between source bit rate and channel coding rate in variable channel states. We compared FGS-JSC method and general single layer CBR(Constant Bit Rate) transmission. In this results, FGS-JSC was appeared better performance than CBR transmission.

• Journal of KIISE:Information Networking
• /
• v.35 no.2
• /
• pp.112-119
• /
• 2008

Today, A matter of concern of home network technology increase. The standard of communication between home network devices are required. IEEE 802.1 AVB(Audio/Video Bridging) specifies transmission method for time-sensitive data between these devices using Ethernet in bridged local area networks. IEEE 802.1 AVB and IEEE 1588 PTP(Precision Time Protocol) have various message type for grandmaster selection and synchronize the devices. These messages bring on complexity protocol. We propose a method that uses Single TimeSync frame in order to the problem. Our proposal is appropriate process complexity and low transmission delay for home network by using the TimeSync frame. Furthermore, after all devices are adjusted to the single TimeSync frame, a resource reservation, a forwarding and queueing rule are needed for a time-sensitive application.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.13 no.2
• /
• pp.176-187
• /
• 2002

The digital Audio Broadcasting (DAB) system adopts Coded OFDM(COFDM) for channel coding. The COFDM is a combined technique of multicarrier transmission(OFDM) and punctured convolutional coding with viterbi error correction. Because the channel coding is an important topic for OFDM systems, this paper proposes a new turbo coded OFDM system that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the frequency selective Rician fading channel and the frequency selective Rayleigh fading channel in conjunction with DAB transmission mode I suitable for the terrestrial single frequency network(SFN) broadcasting. The standard system's performance was improved with the aid of turbo codec.