• Title/Summary/Keyword: Digital codec

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Radio Broadcast Digitalization and a Proposal of USAC as Audio Codec for Digital Radio Broadcast (디지털라디오방송 전환과 오디오 코덱의 USAC 적용 연구)

  • Lee, SangWoon;Lee, Yong Ju;Lee, Taejin;Kang, Kyeongok
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2011.07a
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    • pp.509-512
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    • 2011
  • 최근 국내에 디지털 라디오 방송을 도입하기 위한 실험방송이 진행 중이며, 조만간 디지털 라디오 방식이 결정될 것으로 전망되고 있다. 본 논문에서는 연내 ISO/IEC 국제표준 제정이 예상되는 USAC의 디지털 라디오 방송에의 적용 가능성을 검토하고, 도입의 장점 및 고려사항 등을 제안하고자 한다.

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Digital Watermarking Using MTWC(Multi-Threshods Wavelet Codec) (MTWO 알고리즘을 이용한 디지털 워터마킹)

  • 김동희;장기준;남궁재찬
    • Proceedings of the Korea Multimedia Society Conference
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    • 2003.11a
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    • pp.9-12
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    • 2003
  • 웨이블릿 변환(Wavelet Transform)의 압축 방식 중 최근 압축률에서 뛰어난 성능을 보이는 MTWC 알고리즘의 특성을 이용하여 추출된 중요 계수에만 워터마크를 삽입하는 방법으로 디지털 영상에 시각적으로 보이지 않게 워터마크를 삽입하고, 워터마크된 영상과 PN(Pseudo Noise)code와의 계수차를 이용하여, 위터마크를 추출하는 통계학적 접근 방법을 제시하며, 워터마크가 삽입된 영상에 대한 다양한 공격(Attack)등에 대해서 기존의 방법들과 비교하여 효과적으로 워터마크를 추출할 수 있는 강인함을 실험을 통해 증명해 보였다.

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A Study of the Digital Modulation using DSP (DSP를 이용한 디지털 변조에 관한 연구)

  • 최상권;최진웅;김정국
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2001.06a
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    • pp.89-92
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    • 2001
  • In this paper, as a study of programmable software radio digital communication, we implemented ASK(Amplitude Shift Keying), FSK(Frequency Shift Keying), and PSK(Phase Shift Keying) modulation using programmable software(algorithm) of DSP(Digital Signal Processor). Moreover, it is possible to select one of those three modulation methods by realizing on single DSP. We adopted Motorola DSP56002 and Crystal CS4215(A/D and D/A converter) for our purpose. The DSP56002 is 24-bit and operates 20 MIPS at 40 MHz, and the CS4215 is 16-bit and supports the maximum 50 kHz sampling frequency.

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Implementation of a 4-Channerl ADPCM CODEC Using a DSP (DSP를 사용한 4채널용 ADPCM CODEC의 실시간 구현에 관한 연구)

  • Lee, Ui-Taek;Lee, Gang-Seok;Lee, Sang-Uk
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.22 no.5
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    • pp.29-38
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    • 1985
  • In this paper we have designed and implemented in real time a simple, efficient and flexible AOPCM cosec using a high speed digital processor, NEC 7720. For ADPCM system, we have used an instantaneous adaptive quantizer and a first-order fixed predictor. The software for NEC 7720 has been developed and it was found that the NEC 7720 was capable of performing the entire ADPCAt algorithm for 4 channels in real time as optimizing the program. Computer simulation has born made to investigate a computational accuracr of NEC 7720 and to de-termine necessary parameters for a ADPCM codec. Real telephone speech, RC-shaped Gaussian noise and 1004 Hz tone signal were used for simulation. In simulation, the parameters werc optimized from the computed SNR and the informal listening test. The developed software was tested in real time operation using a hardware emulator for NEC 7720. It took a maximum 23.25$\mu$s to encode one sample and 113.5$\mu$s, including all the necessary 1/0 operations, to encode 4 channels. In the case of decoding process, it took 24.75$\mu$s to decode one sample and 119.5$\mu$s to decode 4 channels.

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Multi-view Video Codec for 3DTV (3DTV를 위한 다시점 동영상 부호화 기법)

  • Bae Jin-Woo;Song Hyok;Yoo Ji-Sang
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.3A
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    • pp.337-344
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    • 2006
  • In this paper, we propose a multi-view video codec for 3DTV system. The proposed algorithm is not only to reduce the temporal and spatial redundancy but also to reduce the redundancy among each view. With these results, we can improve the coding efficiency for multi-view video sequences. In order to reduce the redundancy of each view more efficiently, we define the assembled image(AI) that is generated by the global disparity compensation of each view. In addition, the proposed algorithm is based on MPEG-2 structure so that we can easily implement 3DTV system without changing the conventional 2D digital TV system. Experimental results show that the proposed algorithm performs very well. It also performs better than MPEG-2 simulcast coding method. The newly proposed codec also supports the view scalability, accurate temporal synchronization among multiple views and random access capability in view dimension.

이동 멀티미디어 방송(DMB)에서의 H.264/AVC압축 파라미터 성능연구

  • Sin, Seung-Ho;Kim, Gyeong-Nam;Kim, Tae-Yong
    • Broadcasting and Media Magazine
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    • v.12 no.4
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    • pp.28-39
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    • 2007
  • 다양한 디지털 기술의 발전으로 인하여 방송형태의 이동 멀티미디어 서비스가 다국적으로 제안되고, 국내에서는 이동 멀티미디어 방송 (DMB: Digital Multimedia Broadcasting)을 통하여 야외나 이동시에도 시청이 가능한 방송서비스가 활발해지고 있다. 휴대 및 이동수신 방송 환경에서 비디온 오디오 및 데이터를 포함한 멀티미디어 방송 서비스를 효율적으로 제공하기 위해서는 다양한 장소에서 수신 영상에 대한 품질 확보가 필수적이다. 본 논문에서는 현재 이동 멀티미디어 방송이 비디오 압축방식으로 채택하고 있는 H.264/AVC 압축 파라미터의 성능 연구에 대하여 기술한다. 현재 국내의 위성/지상파 DMB의 경우 비디오의 압축 방법으로 H.264/AVC baseline 1.3의 표준규격을 사용한다. 이러한 비디오 코덱(codec) 이용하여 비디오 영상을 압축할 경우 관련 파라미터(parameter) 조절이 가능한데, 비디오를 압축할 경우 관련 파라미터들을 어떻게 정하느냐에 따라 서로 다른 수신환경에서 압축 효율 및 재생된 비디오의 화질에 많은 영향을 미친다. 따라서 수신 환경에 가장 적합한 비디오 화질을 얻기 위해서는 관련 파라미터 설정이 매우 중요하다. 본 논문에서는 다양한 압축 파라미터들 중 화질에 많은 영향을 미치는 항목을 선정하여, 해당 파라미터의 변화가 재생된 비디오 화질에 미치는 영향을 객관적 평가척도인 PSNR, Bit-rate, 수행시간 등을 이용하여 분석하였다. 또한, 실험 결과를 바탕으로 이동 멀티미디어 방송 환경에서의 H.264 인코더의 적정 압축 파라미터 및 인코더의 성능 개선 방안을 제안한다.

Design of EVRC LSP Codebooks with Korean (한국어에 의한 EVRC LSP 코드북 설계)

  • 이진걸
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.2
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    • pp.167-172
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    • 2002
  • The EVRC (Enhanced Variable Rate Codec) is currently in service as a speech cosec in digital cellular systems in North America and Korea. In the EVRC, the LSP (Line Spectral Pairs) related to energy distribution of speech signals in the frequency domain are coded by weighted split vector quantization. Considering that the LSP codebooks might be trained with the language of the develop country of the codebooks or English, it is expected that codebooks trained with Korean provide the performance improvements in the communication in Korean. In this paper, the EVRC LSP codebooks are designed with korean adopting the LBG algorithm based vector quantization, and the performance improvement of the vector quantization and the accompanying speech quality improvement are demonstrated by spectral distortion, SNR and SegSNR measurements, respectively.

Adaptive Pre-/Post-Filters for NRT-Based Stereoscopic Video Coding

  • Lee, Byung-Tak;Lee, BongHo;Choi, Haechul;Kim, Jin-Soo;Yun, Kugjin;Cheong, Won-Sik;Kim, Jae-Gon
    • ETRI Journal
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    • v.34 no.5
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    • pp.666-673
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    • 2012
  • Non-real-time delivery of stereoscopic video has been considered as a service scenario for 3DTV to overcome the limited bandwidth in the terrestrial digital television system. A hybrid codec combining MPEG-2 and H.264/AVC has been suggested for the compression of stereoscopic video for 3DTV. In this paper, we propose a stereoscopic video coding scheme using adaptive pre-/post-filters (APPF) to improve the quality of 3D video while retaining compatibility with legacy video coding standards. The APPF are applied adaptively to blocks of various sizes determined by the macroblock coding mode and reference frame index. Experiment results show that the proposed method achieves up to 24.86% bit rate savings relative to a hybrid codec of MPEG-2 and H.264/AVC including the inter-view prediction.

Fast Implementation Algorithms for EVRC (EVRC의 고속 구현 알고리듬)

  • 정성교;최용수;김남건;윤대희
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.1
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    • pp.43-49
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    • 2001
  • EVRC (Enhanced Variable Rate Codec) has been adopted as a standard coder for the CDMA digital cellular system in North America and Korea, and known to provide good call quality at 8kbps. In this paper, fast implementation algorithms for EVRC encoder are proposed. The proposed algorithms are based on both efficient pitch detection scheme and fast fixed codebook search algorithm. In the codebook search, computational complexity is reduced down to 70% of the original EVRC by limiting the number of pulse position combination and by using a truncated impulse response. The proposed algorithms enable us to implement the EVRC with much smaller computational works. Also, informal subjective tests confirmed that the difference in the speech quality between the original EVRC and the proposed method was indistinguishable.

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Transmission of Channel Error Information over Voice Packet (음성 패킷을 이용한 채널의 에러 정보 전달)

  • 박호종;차성호
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.4
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    • pp.394-400
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    • 2002
  • In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.