• Title/Summary/Keyword: Codec2

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DCT-domain MPEG-2/H.264 Video Transcoder System Architecture for DMB Services (DMB 서비스를 위한 DCT 기반 MPEG-2/H.264 비디오 트랜스코더 시스템 구조)

  • Lee Joo-Kyong;Kwon Soon-Young;Park Seong-Ho;Kim Young-Ju;Chung Ki-Dong
    • The KIPS Transactions:PartB
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    • v.12B no.6 s.102
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    • pp.637-646
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    • 2005
  • Most of the multimedia contents for DBM services art provided as MPEG-2 bit streams. However, they have to be transcoded to H.264 bit streams for practical services because the standard video codec for DMB is H.264. The existing transcoder architecture is Cascaded Pixel-Domain Transcoding Architecture, which consists of the MPEG-2 dacoding phase and the H.264 encoding phase. This architecture can be easily implemented using MPEG-2 decoder and H.264 encoder without source modifying. However. It has disadvantages in transcoding time and DCT-mismatch problem. In this paper, we propose two kinds of transcoder architecture, DCT-OPEN and DCT-CLOSED, to complement the CPDT architecture. Although DCT-OPEN has lower PSNR than CPDT due to drift problem, it is efficient for real-time transcoding. On the contrary, the DCT-CLOSED architecture has the advantage of PSNR over CPDT at the cost of transcoding time.

An Effective Method to Treat The Boundary Pixels for Image Compression with DWT (DWT를 이용한 영상압축을 위한 경계화소의 효과적인 처리방법)

  • 서영호;김종현;김대경;유지상;김동욱
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.6A
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    • pp.618-627
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    • 2002
  • In processing images using 2 dimensional Discrete Wavelet Transform(2D-DWT), the method to process the pixels around the image boundary may affect the quality of image and the cost to implement in hardware and software. This paper proposed an effective method to treat the boundary pixels, which is apt to implement in hardware and software without losing the quality of the image costly. This method processes the 2-D image as 1-D array so that 2-D DWT is performed by considering the image with the serial-sequential data structure (Serial-Sequential Processing). To show the performance and easiness in implementation of the proposed method, an image compression codec which compresses image and reconstructs it has been implemented and experimented. It included log-scale fried quantizer, but the entropy coder was not implemented. From the experimental results, the proposed method showed the SNR of almost the same SNR(Signal to Noise Ratio) to the Periodic Expansion(PE) method when the compression ratio(excluding entropy coding) of 2:1, 15.3% higher than Symmetric Expansion(SE) method, and 9.3% higher than 0-pixel Padding Expansion(ZPE) method. Also PE method needed 12.99% more memory space than the proposed method. By considering only the compression process, SE and ZPE methods needed additional operations than the proposed one. In hardware implementation, the proposed method in this paper had 5.92% of overall circuit as the control circuit, while SE, PE, and ZPE method has 22%, 21,2%, and 11.9% as the control circuit, respectively. Consequently, the proposed method can be thought more effective in implementing software and hardware without losing any image quality in the usual image processing applications.

HEVC Encoder Optimization using Depth Information (깊이정보를 이용한 HEVC의 인코더 고속화 방법)

  • Lee, Yoon Jin;Bae, Dong In;Park, Gwang Hoon
    • Journal of Broadcast Engineering
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    • v.19 no.5
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    • pp.640-655
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    • 2014
  • Many of today's video systems have additional depth camera to provide extra features such as 3D support. Thanks to these changes made in multimedia system, it is now much easier to obtain depth information of the video. Depth information can be used in various areas such as object classification, background area recognition, and so on. With depth information, we can achieve even higher coding efficiency compared to only using conventional method. Thus, in this paper, we propose the 2D video coding algorithm which uses depth information on top of the next generation 2D video codec HEVC. Background area can be recognized with depth information and by performing HEVC with it, coding complexity can be reduced. If current CU is background area, we propose the following three methods, 1) Earlier stop split structure of CU with PU SKIP mode, 2) Limiting split structure of CU with CU information in temporal position, 3) Limiting the range of motion searching. We implement our proposal using HEVC HM 12.0 reference software. With these methods results shows that encoding complexity is reduced more than 40% with only 0.5% BD-Bitrate loss. Especially, in case of video acquired through the Kinect developed by Microsoft Corp., encoding complexity is reduced by max 53% without a loss of quality. So, it is expected that these techniques can apply real-time online communication, mobile or handheld video service and so on.

Audio Stream Delivery Using AMR(Adaptive Multi-Rate) Coder with Forward Error Correction in the Internet (인터넷 환경에서 FEC 기능이 추가된 AMR음성 부호화기를 이용한 오디오 스트림 전송)

  • 김은중;이인성
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.26 no.12A
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    • pp.2027-2035
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    • 2001
  • In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

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Design of digital decimation filter for sigma-delta A/D converters (시그마-델타 A/D 컨버터용 디지털 데시메이션 필터 설계)

  • Byun, San-Ho;Ryu, Seong-Young;Choi, Young-Kil;Roh, Hyung-Dong;Nam, Hyun-Seok;Roh, Jeong-Jin
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.44 no.2
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    • pp.34-45
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    • 2007
  • Digital decimation filter is inevitable in oversampled sigma-delta A/D converters for the sake of reducing the oversampled rate to Nyquist rate. This paper presented a Verilog-HDL design and implementation of an area-efficient digital decimation filter that provides time-to-market advantage for sigma-delta analog-to-digital converters. The digital decimation filter consists of CIC(cascaded integrator-comb) filter and two cascaded half-band FIR filters. A CSD(canonical signed digit) representation of filter coefficients is used to minimize area and reduce in hardware complexity of multiplication arithmetic. Coefficient multiplications are implemented by using shifters and adders. This three-stage decimation filter is fabricated in $0.25-{\mu}m$ CMOS technology and incorporates $1.36mm^2$ of active area, shows 4.4 mW power consumption at clock rate of 2.8224 MHz. Measured results show that this digital decimation filter is suitable for digital audio decimation filters.

A Study on the Improvements of the Speech Quality by using Distribution Characteristics of LSP parameters in the EVRC(Enhanced Variable Rate Codec) (LSP 파라미터의 분포특성을 이용한 EVRC의 음질개선에 관한 연구)

  • Min, So-Yeon;Na, Deok-Su
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.12 no.12
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    • pp.5843-5848
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    • 2011
  • To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC's basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.

Motion Vector Coding Using Adaptive Motion Resolution (적응적인 움직임 벡터 해상도를 이용한 움직임 벡터 부호화 방법)

  • Jang, Myung-Hun;Seo, Chan-Won;Han, Jong-Ki
    • Journal of Broadcast Engineering
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    • v.17 no.1
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    • pp.165-178
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    • 2012
  • In most conventional video codecs, such as MPEG-2 and MPEG-4, inter coding is performed with the fixed motion vector resolution. When KTA software was developed, resolution for MVs can be selected in each slice. Although KTA codec uses a variety of resolutions for ME, the selected resolution is applied over the entire pixels in the slice and the statistical property of the local area is not considered. In this paper, we propose an adaptive decision scheme for motion vector resolution which depends on region, where MV search area is divided to multiple regions according to the distance from PMV. In each region, the assigned resolution is used to estimate MV. Each region supports different resolution for ME from other regions. The efficiency of the proposed scheme is affected from threshold values to divide the search area and the entropy coding method to encode the estimated MV. Simulation results with HM3.0 which is the reference software of HEVC show that the proposed scheme provides bit rate gains of 0.9%, 0.6%, and 2.9% in Random Access, Low Delay with B picture, and Low Delay with P picture structures, respectively.

Point Cloud Video Codec using 3D DCT based Motion Estimation and Motion Compensation (3D DCT를 활용한 포인트 클라우드의 움직임 예측 및 보상 기법)

  • Lee, Minseok;Kim, Boyeun;Yoon, Sangeun;Hwang, Yonghae;Kim, Junsik;Kim, Kyuheon
    • Journal of Broadcast Engineering
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    • v.26 no.6
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    • pp.680-691
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    • 2021
  • Due to the recent developments of attaining 3D contents by using devices such as 3D scanners, the diversity of the contents being used in AR(Augmented Reality)/VR(Virutal Reality) fields is significantly increasing. There are several ways to represent 3D data, and using point clouds is one of them. A point cloud is a cluster of points, having the advantage of being able to attain actual 3D data with high precision. However, in order to express 3D contents, much more data is required compared to that of 2D images. The size of data needed to represent dynamic 3D point cloud objects that consists of multiple frames is especially big, and that is why an efficient compression technology for this kind of data must be developed. In this paper, a motion estimation and compensation method for dynamic point cloud objects using 3D DCT is proposed. This will lead to switching the 3D video frames into I frames and P frames, which ensures higher compression ratio. Then, we confirm the compression efficiency of the proposed technology by comparing it with the anchor technology, an Intra-frame based compression method, and 2D-DCT based V-PCC.

A study of next generation OpenCable systems for Ultra-High Definition television broadcasting (초 고화질 텔레비전 방송을 위한 차세대 오픈 케이블 방식에 대한 연구)

  • Cho, Chang-Yeon;Heo, Jun;Kim, Joon-Tae
    • Journal of Broadcast Engineering
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    • v.14 no.2
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    • pp.228-237
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    • 2009
  • This paper examines the potential of Ultra-High Definition TV (UD-TV) broadcasting transmission systems beyond HD-TV over cable channel. Firstly, we analyze the trend of TOV(Threshold of Visibility) by extending the OpenCable (J.83 Annex B) system 256QAM which is the standard of Korean and American cable television transmission to 1024QAM, and realize that the OpenCable 1024QAM has nearly 30% higher data rate than 256QAM at the expense of impractically higher TOV (Threshold of Visibility). To achieve practical TOV, we control code rates of inner convolutional coder and replace turbo coder in forward error correction (FEC) part, thereby analyzing the best performance of the OpenCable systems having conventional FEC. In that result, it is necessary to modify conventional FEC of the OpenCable system to achieve under 31.5dB TOV. Moreover we study the potential of UD-TV transmission via two or more TV channels, so called channel bonding, through the Shannon capacity in 6MHz channel and the relationship with next generation A/V codec technologies.

Design and Optimization of Mu1ti-codec Video Decoder using ASIP (ASIP를 이용한 다중 비디오 복호화기 설계 및 최적화)

  • Ahn, Yong-Jo;Kang, Dae-Beom;Jo, Hyun-Ho;Ji, Bong-Il;Sim, Dong-Gyu;Eum, Nak-Woong
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.48 no.1
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    • pp.116-126
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    • 2011
  • In this paper, we present a multi-media processor which can decode multiple-format video standards. The designed processor is evaluated with optimized MPEG-2, MPEG-4, and AVS (Audio video standard). There are two approaches for developing of real-time video decoders. First, hardware-based system is much superior to a processor-based one in execution time. However, it takes long time to implement and modify hardware systems. On the contrary, the software-based video codecs can be easily implemented and flexible, however, their performance is not so good for real-time applications. In this paper, in order to exploit benefits related to two approaches, we designed a processor called ASIP(Application specific instruction-set processor) for video decoding. In our work, we extracted eight common modules from various video decoders, and added several multimedia instructions to the processor. The developed processor for video decoders is evaluated with the Synopsys platform simulator and a FPGA board. In our experiment, we can achieve about 37% time saving in total decoding time.