• Title/Summary/Keyword: real-time encoding and decoding

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A Parallel Hardware Architecture for H.264/AVC Deblocking Filter (H.264/AVC를 위한 블록현상 제거필터의 병렬 하드웨어 구조)

  • Jeong, Yong-Jin;Kim, Hyun-Jip
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.43 no.10 s.352
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    • pp.45-53
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    • 2006
  • In this paper, we proposed a parallel hardware architecture for deblocking filter in K264/AVC. The deblocking filter has high efficiency in H.264/AVC, but it also has high computational complexity. For real time video processing, we chose a two 1-D parallel filter architecture, and tried to reduce memory access using dual-port SRAM. The proposed architecture has been described in Verilog-HDL and synthesized on Hynix 0.25um CMOS Cell Library using Synopsys Design Compiler. The hardware size was about 27.3K logic gates (without On-chip Memory) and the maximum operating frequency was 100Mhz. It consumes 258 clocks to process one macroblock, witch means it can process 47.8 HD1080P(1920pixel* 1080pixel) frames per second. It seems that it can be used for real time H.264/AVC encoding and decoding of various multimedia applications.

Conversational Quality Measurement System for Mobile VoIP Speech Communication (모바일 VoIP 음성통신을 위한 대화음질 측정 시스템)

  • Cho, Jae-Man;Kim, Hyoung-Gook
    • The Journal of The Korea Institute of Intelligent Transport Systems
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    • v.10 no.4
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    • pp.71-77
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    • 2011
  • In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

Real-time Implementation of Variable Transmission Bit Rate Vocoder Integrating G.729A Vocoder and Reduction of the Computational Amount SOLA-B Algorithm Using the TMS320C5416 (TMS320C5416을 이용한 G.729A 보코더와 계산량 감소된 SOLA-B 알고리즘을 통합한 가변 전송율 보코더의 실시간 구현)

  • 함명규;배명진
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.40 no.6
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    • pp.84-89
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    • 2003
  • In this paper, we real-time implemented to the TMS320C5416 the vocoder of variable bit rate applied the SOLA-B algorithm by Henja to the ITU-T G.729A vocoder of 8kbps transmission rate. This proposed method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. At this time, we bandied that the interval of cross correlation function if skipped every 3 sample for decreasing the computational amount of SOLA-B algorithm. The real-time implemented vocoder of C.729A and SOLA-B algorithm is represented the complexity of maximum that is 10.2MIPS in encoder and 2.8MIPS in decoder of 8kbps transmission rate. Also, it is represented the complexity of maximum that is 18.5MIPS in encoder and 13.1MIPS in decoder of 6kbps, it is 18.5MIPS in encoder and 13.1MIPS in decoder of 4kbps. The used memory is about program ROM 9.7kwords, table ROM 4.5kwords, RAM 5.1 kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, for evaluation of speech quality of the vocoder of real-time implemented variable bit rate, it is estimated the MOS score of 3.69 in 4kbps.

Wavelet Packet Image Coder Using Coefficients Partitioning For Remote Sensing Images (위성 영상을 위한 계수분할 웨이블릿 패킷 영상 부호화 알고리즘에 관한 연구)

  • 한수영;조성윤
    • Korean Journal of Remote Sensing
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    • v.18 no.6
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    • pp.359-367
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    • 2002
  • In this paper, a new embedded wavelet packet image coder algorithm is proposed for an effective image coder using correlation between partitioned coefficients. This new algorithm presents parent-child relationship for reducing image reconstruction error using relations between individual frequency sub-bands. By parent-child relationship, every coefficient is partitioned and encoded for the zerotree data structure. It is shown that the proposed wavelet packet image coder algorithm achieves low bit rates and rate-distortion. It also demonstrates higher PSNR under the same bit rate and an improvement in image compression time. The perfect rate control is compared with the conventional method. These results show that the encoding and decoding processes of the proposed coder are simpler and more accurate than the conventional ones for texture images that include many mid and high-frequency elements such as aerial and satellite photograph images. The experimental results imply the possibility that the proposed method can be applied to real-time vision system, on-line image processing and image fusion which require smaller file size and better resolution.

Implementation of the TMS320C6701 DSP Board for Multichannel Audio Coding (멀티채널 오디오 부호화를 위한 TMS320C6701 DSP 보드 구현)

  • 장대영;홍진우;곽진석
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 1999.11a
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    • pp.199-203
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    • 1999
  • This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

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Cross-layer Design of Packet Scheduling for Real-Time Multimedia Streaming (실시간 멀티미디어 스트리밍을 위한 계층 통합 패킷 스케줄링 기법)

  • Hong, Sung-Woo;Won, You-Jip
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.11B
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    • pp.1151-1168
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    • 2009
  • Improving packet loss does not necessarily coincide with the improvement in user perceivable QoS because each frame carries different degree of importance. We propose Significance-aware packet scheduling (SAPS) to maximize user perceivable QoS. SAPS carries out two fundamental issues of packet scheduling: "What to transmit" and "When to transmit?" To adapt to the available bandwidth, it is necessarily to transmit the subset of the data packets if the entire set of packets can not be transmitted. "Packet Significance" quantifies the importance of the frame by elaborately incorporating frames' dependency. Greedy approach is used in selecting packets and transmission schedule is determined based on the Packet Significance. The proposed scheme is tested using publicly available MPEG-4 video clips. Decoding engine is embedded in the simulation software and user perceivable QoS is exposeed in termstermiSNR. Throughout the simulation based experiment, the performance of the proposed scheme is compared two other schemes: Size-based packet scheduling and Bit-rate based best effort packet scheduling. SAPS successfully incorporates the semantics of a packet and improves user perceivable QoS significantly. It successfully provides unequal protection to more important packets.

Fast Game Encoder Based on Scene Descriptor for Gaming-on-Demand Service (주문형 게임 서비스를 위한 장면 기술자 기반 고속 게임 부호화기)

  • Jeon, Chan-Woong;Jo, Hyun-Ho;Sim, Dong-Gyu
    • Journal of Korea Multimedia Society
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    • v.14 no.7
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    • pp.849-857
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    • 2011
  • Gaming on demand(GOD) makes people enjoy games by encoding and transmitting game screen at a server side, and decoding the video at a client side. In this paper, we propose a fast game video encoder for multiple users over network with low-powered devices. In the proposed system, the computational complexity of game encoders is reduced by using scene descriptors, which consists of an object motion vector, global motion, and scene change. With additional information from game engines, the proposed encoder does not need to perform various complexity processes such as motion estimation and ratedistortion optimization. The motion estimation and rate-distortion optimization skipped by scene descriptors. We found that the proposed method improved 192 % in terms of FPS, compared with x264 software. With partial assembly code, we also improved coding speed by 86 % in terms of FPS. We found that the proposed fast encoder could encode over 60 FPS for real-time GOD applications.

Design of FPGA Camera Module with AVB based Multi-viewer for Bus-safety (AVB 기반의 버스안전용 멀티뷰어의 FPGA 카메라모듈 설계)

  • Kim, Dong-jin;Shin, Wan-soo;Park, Jong-bae;Kang, Min-goo
    • Journal of Internet Computing and Services
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    • v.17 no.4
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    • pp.11-17
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    • 2016
  • In this paper, we proposed a multi-viewer system with multiple HD cameras based AVB(Audio Video Bridge) ethernet cable using IP networking, and FPGA(Xilinx Zynq 702) for bus safety systems. This AVB (IEEE802.1BA) system can be designed for the low latency based on FPGA, and transmit real-time with HD video and audio signals in a vehicle network. The proposed multi-viewer platform can multiplex H.264 video signals from 4 wide-angle HD cameras with existed ethernet 1Gbps. and 2-wire 100Mbps cables. The design of Zynq 702 based low latency to H.264 AVC CODEC was proposed for the minimization of time-delay in the HD video transmission of car area network, too. And the performance of PSNR(Peak Signal-to-noise-ratio) was analyzed with the reference model JM for encoding and decoding results in H.264 AVC CODEC. These PSNR values can be confirmed according the theoretical and HW result from the signal of H.264 AVC CODEC based on Zynq 702 the multi-viewer with multiple cameras. As a result, proposed AVB multi-viewer platform with multiple cameras can be used for the surveillance of audio and video around a bus for the safety due to the low latency of H.264 AVC CODEC design.

Parallelization Method of Slice-based video CODEC (슬라이스 기반 비디오 코덱 병렬화 기법)

  • Nam, Jung-Hak;Ji, Bong-Il;Jo, Hyun-Ho;Sim, Dong-Gyu;Cho, Dae-Sung
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.47 no.6
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    • pp.48-56
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    • 2010
  • Recently, we need to dramatically speed up real-time video encoding and decoding on mobile devices because complexity of video CODEC is significantly increasing along with the demand for multimedia service of high-quality and high-definition videos by users. A variety of research is conducted for parallelism of video processing using newly developed multi-core platforms. In this paper, we propose a method of parallelism based on slice partition of video compression CODEC. We propose a novel concept of a parallel slice for parallelism and propose a new coding order to be adequate to the parallel slice which keeps high coding efficiency. To minimize synchronization time of multiple parallel slices, we also propose a synchronization method to determinate whether the parallel slice could be independently decoded or not. Experimental results shows that we achieved 27.5% (40.7%) speed-up by parallelism with bit-rate increase of 3.4% (2.7%) for CIF sequences (720p sequences) by implementing the proposed algorithm on the H.264/AVC.

A mechanism for end-to-end secure communication in heterogeneous tactical networks (이기종 전술통신망 종단간 암호화 통신을 위한 메커니즘)

  • Park, Cheol-Yong;Kim, Ki-Hong;Ryou, Jae-Cheol
    • Journal of the Korea Institute of Information Security & Cryptology
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    • v.24 no.4
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    • pp.625-634
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    • 2014
  • Tactical networks is being operated in configuration that consisting of a variety of characteristics communication equipments and heterogeneous networks. In this configurations, end-to-end communication can be achieved using interworking gateway for converting the data format of the network and using encryption algorithm of the networks. The use of mechanism results in a problem that secure data cannot be transferred directly, reprocessing and processing delay of communication in heterogeneous tactical networks. That is, for encoding and decoding of data, the decryption of encrypted data and re-encryption processing must be required at the gateway between different networks. In this paper proposes to mechanism for end-to-end secure communication in heterogeneous tactical networks. Using the proposed method, end-to-end secure communication between heterogeneous tactical networks(PSTN-UHF networks) which removes the necessity of a gateway for converting data into data formats suitable for network to remove a transmission delay factor and enable real-time voice and data communication and achieve end-to-end security for heterogeneous tactical networks. we propose a novel mechanism for end-to-end secure communication over PSTN and UHF networks and evaluate against the performance of conventional mechanism. Our proposal is confirmed removal of security vulnerabilities, end-to-end secure communication in heterogeneous tactical networks.