• Title/Summary/Keyword: Digital Audio

Search Result 626, Processing Time 0.029 seconds

Synchronization Method and Link Level Performance of DMB System A considering HPA Nonlineariry (HPA 비선형성을 고려한 DMB 시스템 A의 링크레벨 성능 및 동기화 기법)

  • Park SungHo;Cha Insuk;Chang KyungHi
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.30 no.6A
    • /
    • pp.488-498
    • /
    • 2005
  • The DAB(Digital Audio Broadcasting) service which is based on the Eureka-147 of Europe is developed to DMB(Digital Multimedia Broadcasting) service that is divided into Terrestrial DMB and Satellite DMB. The Satellite DMB is a new broadcasting service, which will service multi-channel multimedia broadcasting by the portable receiver or the vehicle receiver. In this paper, we consider that link level performance of satellite DMB system A which is based on the COFDM(Coded Orthogonal Division Multiplexing). It uses the OFDM method which is sensitive to nonlinearity, so we analyze the effect of the HPA(High Power Amplifier) nonlinearity. And then we define the appropriate back-off value by performing the link level simulation considering back-off effect. Also we consider the effect of frequency and time offset, and then confirm the overall link level performance by analyzing and verifying a suitable synchronization method for satellite DMB system A.

A Study on Transport Stream Analysis and Parsing Ability Enhancement in Digital Broadcasting and Service (디지털 방송 서비스에서 트랜스포트 스트림 분석 및 파싱 능력 향상에 관한 연구)

  • Kim, Jang-Won
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
    • /
    • v.10 no.6
    • /
    • pp.552-557
    • /
    • 2017
  • Wire, wireless digital broadcasting has sharply expanded with the birth of high definition TV since 2010, the use of duplex contents as well as simplex contents has rapidly increased. Currently, our satellite communications system adopted DVB by European digital broadcasting standardization organization as a standard of domestic data broadcasting, the method how to use selective contents has been studied variously according to the development of IPTV. Digital broadcasting utilizes the method using Transport Stream Packet(TSP) by the way of multiplexing of information in order to send multimedia information such as video, audio and data of MPEG-2, this streams include detail information on TV guide and program as well as video and audio information. In order to understand these data broadcasting system, this study realized TS analyzer that divides transport stream (TS) by packet in Linux environment, analyzes and prints by function, it can help the understanding of TS, the enhancement of stream parsing ability.

MPEG-4 BIFS Optimization for Interactive T-DMB Content (지상파 DMB 컨텐츠의 MPEG-4 BIFS 최적화 기법)

  • Cha, Kyung-Ae
    • Journal of Korea Society of Industrial Information Systems
    • /
    • v.12 no.1
    • /
    • pp.54-60
    • /
    • 2007
  • The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality multimedia content to the mobile environment. The system adopts the MPEG-4 standard for the main video, audio and other media format. For providing interactive contents, it also adopts the MPEG-4 scene description that refers to the spatio-temporal specifications and behaviors of individual objects. With more interactive contents, the scene description also needs higher bitrate. However, the bandwidth for allocating meta data, such as scene description is restrictive in the mobile environment. On one hand, the DMB terminal renders each media stream according to the scene description. Thus the binary format for scene(BIFS) stream corresponding to the scene description should be decoded and parsed in advance when presenting media data. With this reasoning, the transmission delay of the BIFS stream would cause the delay in transmitting whole audio-visual scene presentations, although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique in adapting the BIFS stream into the expected bitrate without any waste in bandwidth and avoiding transmission delays inthe initial scene description for interactive DMB content.

  • PDF

An Efficient PN Sequence Embedding and Detection Method for High Quality Digital Audio Watermarking (고음질 디지털 오디오 워터마킹을 위한 효율적인 PN 시퀸스 삽입 및 검출 방법)

  • 김현욱;오현오;김연정;윤대희
    • Journal of Broadcast Engineering
    • /
    • v.6 no.1
    • /
    • pp.21-31
    • /
    • 2001
  • In the PN-sequence based audio watermarking system, the PN sequence is shaped by a filter derived from the psychoacoustic model to increase robustness and inaudibility The psychoacoustic model calculated in each audio segment, however, requires heavy computational loads. In this paper, we propose an efficient watermarking system adopting a fixed-shape perceptual filter that substitutes psychoacoustic model derived filter. The proposed filter can shape the PN-sequence to be inaudible and enable to embed the robust watermark in a simple manner. Moreover, we propose an anchitecture for the PN-sequence compensation fitter In the watermark detecter to increase correlation between the watermark and the PN-sequence. With the proposed architecture, the blind watermark detection performance has been enhanced.

  • PDF

Authoring Tool of Musical Slide Show MAF Contents

  • Sabirin Muhammad Syah Houari;Kim Mun-Churl
    • Proceedings of the Korean Society of Broadcast Engineers Conference
    • /
    • 2006.11a
    • /
    • pp.289-295
    • /
    • 2006
  • The Musical Slide Show MAF, which currently being standardized by MPEG, conveys the concept of combining several established standard technologies in a single file format. It defines the format of packing up MP3 audio data, along with MPEG-7 Simple Metadata Profile and MPEG-21 Digital Item Declaration metadata; with JPEC images and optional text, and synchronizes them all together to create a slideshow of JPEC image data associated to MP3 audio data during the audio playback. The implementation of Musical Slide Show MAF can be a music karaoke file where users can sing along while listening to the music, view the JPEG slideshow and reading the lyrics; or a story-telling file where users can listen to the narrated story by looking at the related illustration slideshow of the story In this paper we present the tool to producing the Musical Slide Show MAF contents. Regardless the knowledge of user on the MAF file format, the authoring tool simplify the manner of packaging several multimedia contents into single file.

  • PDF

A Hybrid Audio ${\Delta}{\Sigma}$ Modulator with dB-Linear Gain Control Function

  • Kim, Yi-Gyeong;Cho, Min-Hyung;Kim, Bong-Chan;Kwon, Jong-Kee
    • ETRI Journal
    • /
    • v.33 no.6
    • /
    • pp.897-903
    • /
    • 2011
  • A hybrid ${\Delta}{\Sigma}$ modulator for audio applications is presented in this paper. The pulse generator for digital-to-analog converter alleviates the requirement of the external clock jitter and calibrates the coefficient variation due to a process shift and temperature changes. The input resistor network in the first integrator offers a gain control function in a dB-linear fashion. Also, careful chopper stabilization implementation using return-to-zero scheme in the first continuous-time integrator minimizes both the influence of flicker noise and inflow noise due to chopping. The chip is implemented in a 0.13 ${\mu}m$ CMOS technology (I/O devices) and occupies an active area of 0.37 $mm^2$. The ${\Delta}{\Sigma}$ modulator achieves a dynamic range (A-weighted) of 97.8 dB and a peak signal-to-noise-plus-distortion ratio of 90.0 dB over an audio bandwidth of 20 kHz with a 4.4 mW power consumption from 3.3 V. Also, the gain of the modulator is controlled from -9.5 dB to 8.5 dB, and the performance of the modulator is maintained up to 5 nsRMS external clock jitter.

A Fast IFFT Algorithm for IMDCT of AAC Decoder (AAC 디코더의 IMDCT를 위한 고속 IFFT 알고리즘)

  • Chi, Hua-Jun;Kim, Tae-Hoon;Park, Ju-Sung
    • The Journal of the Acoustical Society of Korea
    • /
    • v.26 no.5
    • /
    • pp.214-219
    • /
    • 2007
  • This paper proposes a new IFFT(Inverse Fast Fourier Transform) algorithm, which is proper for IMDCT(Inverse Modified Discrete Cosine Transform) of MPEG-2 AAC(Advanced Audio Coding) decoder. The $2^n$(N-point) type IMDCT is the most powerful among many IMDCT algorithms, however it includes IFFT that requires many calculation cycles. The IFFT used in $2^n$(N-point) type IMDCT employ the bit-reverse data arrangement of inputs and N/4-point complex IFFT to reduce the calculation cycles. We devised a new data arrangement method of IFFT input and $N/4^{n+1}$-type IFFT and thus we can reduce multiplication cycles, addition cycles, and ROM size.

A study on adaptive equalization for OFDM system over Multipath fading channels (다중 경로하에서의 OFDM 시스템을 위한 적응등화에 대한 연구)

  • 이승호;유종엽;우대호;변윤식
    • Proceedings of the IEEK Conference
    • /
    • 2000.09a
    • /
    • pp.229-232
    • /
    • 2000
  • Orthogonal frequency division multiplexing(OFDM) has meanwhile become part of several telecommunicati ons standards, such as satellite and terrestrial digital audio broadcasting(DAB), digital terrestrial TV broad casting(DVB), asymmetric digital subscriber line(ADSL) for high-bit-rate digital subscriber services on twisted-pair channels, and broadband indoor wireless systems. In his paper, we show that OFDM signals contain sufficient structure to accomplish blind channel estimation using second order statistics only. This method doesn't require redundancy as cp in transmitter. And the result is compared with PSAM channel estimation as least square, linear minimum mean square, singular value decomposition.

  • PDF

A Digital Convergence Platform based on the MPEG-21 Multimedia Framework (MPEG-21기반 디지털 컨버젼스 플랫폼 기술)

  • Oh, Hwa-Yong;Kim, Dong-Hwan;Lee, Eun-Seo;Chang, Tae-Gyu
    • The Transactions of The Korean Institute of Electrical Engineers
    • /
    • v.56 no.5
    • /
    • pp.987-989
    • /
    • 2007
  • This paper describes a digital convergence platform(DCP) which is implemented based on the MPEG-21 multimedia framework. DCP is a newly proposed solution in this research for the convergence service of future home multimedia environment. DCP is a common platform designed to have the feature of reconfigurability, by means of S/W. which supports to run diverse digital multimedia services. A distributed peer to peer service and transaction model is also a new feature realized in DCP using the MPEG-21 multimedia framework. A prototype DCP is implemented to verify its functions of multimedia service and transactions. The developed DCPs are networked with IP clustering storage systems for the distributed service of multimedia. Successful streaming services of the MPEG-2/4 video and audio are verified with the implemented test-bed system of DCP.

Efficient Native Processing Modules for Interactive DTV Middleware Based on the Small Footprint Set-Top Box

  • Shin, Sang-Myeong;Im, Dong-Gi;Jung, Min-Soo
    • Journal of Korea Multimedia Society
    • /
    • v.9 no.12
    • /
    • pp.1617-1627
    • /
    • 2006
  • The concept of middleware for digital TV receivers is not new one. Using middleware for digital TV development has a number of advantages. It makes it easier for manufacturers to hide differences in the underlying hardware. It also offers a standard platform for application developers. Digital TV middleware enables set-top boxes(STBs) to run video, audio, and applications. The main concern of digital TV middleware is now to reduce its memory usage because most STBs in the market are small footprint. In this paper, we propose several ideas about how to reduce the required memory size on the runtime area of DTV middleware using a new native process technology. Our proposed system has two components; the Efficient Native Process Module, and Enhanced Native Interface APIs for concurrent native modules. With our approach, the required memory reduced from 50% up to 75% compared with the traditional approach. It can be suitable for low end STBs of very low hardware limitation.

  • PDF