• Title/Summary/Keyword: Multichannel Audio

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Image Enhancement Techniques for MPEG-4 (MPEG-4 영상의 화질 개선에 관한 연구)

  • 김태근;신정호;백준기
    • Journal of Broadcast Engineering
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    • v.2 no.2
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    • pp.169-181
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    • 1997
  • In this paper, we propose and discuss about image enhancement techniques for MPEG-4. which represents very low bit-rate, content-based. and object-based hierarchical audio-visual coding standard. The proposed enhancement technique removes undesired artifacts arising in the compression procedure and increase resolution in both spatial and temporal domains. In order to remove undesired artifacts. we divide the MPEG-4 video algorithm in two parts: MPEG-2 like part and the new part. For removing artifacts caused by the first part. we adopt the conventional blocking artifacts algorithm developed for MPEG-2. On the other hand for removing artifacts caused by the second part. we provide a new degradation model. and propose the corresponding image restoration method. For increasing resolution of the MPEG-4 images, we propose a general framework of multichannel image interpolation process. which includes both spatial and temporal interpolations. As the MPEG-4 standard is under development. various sophisticated techniques are considered. but research on image enhancement techniques is relatively underestimated. By this reason. additional image enhancement techniques will become very important issue in realization phase of MPEG-4.

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On the Principles and Applications of Wave Field Synthesis (WFS의 원리와 활용에 관하여)

  • Yoo, Jae-Hyoun;Shim, Hwan;Chung, Hyun-Joo;Sung, Koeng-Mo;Kang, Kyeong-Ok
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.8
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    • pp.688-696
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    • 2009
  • There are many studies on Wave Field Synthesis(WFS) which provides better presence and spaciousness than conventional discrete multichannel audio reproduction methods. However, it has several problems such as the listener-enclosing loudspeaker array and pre-authorized object-based source signal, so it is not widely used except in large-scale listening rooms. This paper presents a method which utilizes the merit of WFS in small listening rooms such as a living room.

Analysis on Protection Ratio of IBAC DAB System for Co-Channel FM Interferer (동일채널 FM 간섭원에 대한 IBAC DAB 시스템의 혼신 보호비 분석)

  • Jeong, Young-Ho;Park, So-Ra;Kim, Geon;Lee, Hyun;Lee, Soo-In
    • Journal of Broadcast Engineering
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    • v.5 no.2
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    • pp.199-210
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    • 2000
  • The IBAC (In-Band Adjacent-Channel) DAB (Digital Audio Broadcasting) system is to provide multichannel CD quality audio services and multimedia data services including text and picture in FM band (88~105 MHz). As the FM band is being used by the existing analog radio broadcasting, there must he an analysis of the interference effect between IBAC DAB and analog FM signal. Therefore, the protection ratio should be evaluated to verify the system compatibility and allocate the new IBAC DAB channel in FM band. In this paper, among the three types of interferences, FM-to-DAB, DAB-to-FM and DAB-to-DAB, that can be occurred, the Protection ratio of IBAC DAB system for co-channel FM interferer is analyzed by modeling the FM interferer and considering the multipath fading channel. The simulation results show that IBAC DAB system has far better sensitivity than Eureka 147 and needs a relatively high protection ratio for co-channel FM interferer, because of its narrow bandwidth, about one third of that of Eureka 147.

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Sound event detection based on multi-channel multi-scale neural networks for home monitoring system used by the hard-of-hearing (청각 장애인용 홈 모니터링 시스템을 위한 다채널 다중 스케일 신경망 기반의 사운드 이벤트 검출)

  • Lee, Gi Yong;Kim, Hyoung-Gook
    • The Journal of the Acoustical Society of Korea
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    • v.39 no.6
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    • pp.600-605
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    • 2020
  • In this paper, we propose a sound event detection method using a multi-channel multi-scale neural networks for sound sensing home monitoring for the hearing impaired. In the proposed system, two channels with high signal quality are selected from several wireless microphone sensors in home. The three features (time difference of arrival, pitch range, and outputs obtained by applying multi-scale convolutional neural network to log mel spectrogram) extracted from the sensor signals are applied to a classifier based on a bidirectional gated recurrent neural network to further improve the performance of sound event detection. The detected sound event result is converted into text along with the sensor position of the selected channel and provided to the hearing impaired. The experimental results show that the sound event detection method of the proposed system is superior to the existing method and can effectively deliver sound information to the hearing impaired.

Research of packetizing method for efficient transmission of multichannel audio on T-DMB environment (지상파 DMB 환경에서 효율적으로 멀티채널 오디오를 전송하기 위한 패킷화 방법 연구)

  • Lee, Yong-Ju;Seo, Jeong-Il;Beack, Seung-Kwon;Kang, Kyeong-Ok;Lim, Jong-Soo
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2008.11a
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    • pp.239-242
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    • 2008
  • 지상파 DMB는 이동 환경에서 QVGA 급의 영상과 스테레오 오디오를 제공하는 방송 서비스로서 2005년 12월부터 본격적으로 서비스되고 있는데, 최근에는 DMB 환경에서 고품질의 영상과 오디오를 제공하려는 기술에 대한 연구가 이루어지고 있다. 지상파 DMB 환경에서 고품질의 영상 또는 오디오를 제공하기 위해서는 기존의 DMB 서비스에 추가적인 데이터들을 전송하는 것이 필요한데, 하나의 지상파 DMB 방송 채널에 할당되는 전송 비트율이 높지 않다는 점을 감안하면, 이러한 추가적인 데이터들을 효율적으로 전송하는 것이 서비스의 상용화 입장에서는 중요한 요소가 될 수 있다. 본 논문에서는 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하고자 할 때, 추가적으로 전송되어야 하는 부가정보 스트림의 효율적인 전송을 위한 패킷화 방법을 제안한다. 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하기 위한 부가정보 스트림은 일반 오디오 스트림과 마찬가지로 프레임 단위로 생성이 되며, 약 12kbps의 비트율을 가진다. 그러나, 부가정보 스트림을 지상파 DMB 환경에서 전송하기 위하여, MPEG-2 TS로 패킷화하여 전송하게 되면, 부가정보 스트림의 비트율보다 훨씬 높은 약 32kbps의 전송율을 가지게 된다. 본 연구에서는 이와 같은 문제점을 해결하기 위하여, 멀티채널 오디오 서비스를 위해 필요한 부가정보 스트림의 비트율을 분석하고, 이를 바탕으로 하나의 TS 패킷에 하나 이상의 부가정보 프레임을 포함하여 전송하는 방법을 제안한다. 제안한 방법의 성능 검증을 위해 제안한 방법에 따라 하나 이상의 부가정보 프레임을 하나의 TS 패킷에 포함하여 패킷화하는 것을 시뮬레이션하고, 그 결과를 제시하였다.

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A Perception Based Active Matrix Decoder with Virtual Source Location Information (가상 음원 위치 정보를 이용한 능동 메트릭스 디코더)

  • Moon, Han-Gil
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.47 no.5
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    • pp.18-24
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    • 2010
  • In this paper, a new matrix decoding system using vector based Virtual Source Location Information (VSLI) is proposed as an alternative to the conventional Dolby Pro logic II/IIx system for reconstructing multi-channel output signals from matrix encoded two channel signals, Lt/Rt. This new matrix decoding system is composed of passive decoding part and active part. The passive part makes crude multi-channel signals using linear combination of the two encoded signals(Lt/Rt) and the active part enhances each channel regarding to the virtual source which is emergent in each inter channel. Since the virtual sources are related to the perceptual sound images in virtual sound field, the reconstructed multi-channel sound results in good dynamic perception and stable image localization. Moreover, the good channel separation is maintained with nonlinear trigonometric enhancing function.