• Journal of Broadcast Engineering
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• v.18 no.2
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• pp.204-214
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• 2013

Recently various research have been conducted relating to the implementation of signal processing or communication system by software using the massively parallel processing capability of the GPU. In this work, we focus on reducing software simulation time of 2K/8K FFT in DVB-T by using GPU. we estimate the processing time of the DVB-T system, which is one of the standards for DTV transmission, by CPU. Then we implement the FFT processing by the software using the NVIDIA's massively parallel GPU processor. In this paper we apply stream process method to reduce the overhead for data transfer between CPU and GPU, coalescing method to reduce the global memory access time and data structure design method to maximize the shared memory usage. The results show that our proposed method is approximately 20~30 times as fast as the CPU based FFT processor, and approximately 1.8 times as fast as the CUFFT library (version 2.1) which is provided by the NVIDIA when applied to the DVB-T 2K/8K mode FFT.

• Journal of Broadcast Engineering
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• v.15 no.2
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• pp.205-216
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• 2010

In this paper, a method of MPEG-2 Transport Stream (TS) multiplexing for Ultra HDTV (UHDTV) and its design and implementation as a SW tool is described. In practice, UHD video may be divided into several HD videos and each video is encoded in parallel. Therefore, it is necessary to synchronize and multiplex multiple bitstreams encoding each HD video for transmitting and storing UHD video. In this paper, it is assumed that 4 HD videos partitioning a UHD spatially are encoded as H.264/AVC and two 5.0 channel audios are encoded by AC-3. Therefore, 4 H.264/AVC elementary streams (ESs) and 2 AC-3 ESs is mainly considered in the TS multiplexing of UHD. For the carriage of H.264/AVC and AC-3 over MPEG-2 TS, PES packetization and TS multiplexing are designed and implemented based on the extended specification of the MPEG-2 Systems and ATSC (Digital audio compressed standard), respectively. The implemented UHD TS multiplexing tool emulates real time HW operation in the time unit corresponding to the duration of one TS packet transmission in a given TS rate. In particular, in order to satisfy the timing model, the buffers defined in the TS System Target Decoder (T-STD) are monitored and their statuses are considered in the scheduling of TS multiplexing. For UHD multiplexing, two kinds of multiplexing structures, which are UHD re-multiplexing and UHD program multiplexing, are implemented and their strength and weakness are investigated. The developed UHD TS multiplexing tool is tested and verified in terms of the syntax and semantics conformance and functionalities by using a commercial analyzer and real-time presentation tools.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.100-110
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• 2001

In this paper, we investigated several buffer management schemes for the design of shared-memory type ATM switches, which can enhance the utilization of switch resources and can support quality-of-service (QoS) functionalities. Our results show that dynamic threshold (DT) scheme demonstrate a moderate degree of robustness close to pushout(PO) scheme, which is known to be impractical in the perspective of hardware implementation, under various traffic conditions such as traffic loads, burstyness of incoming traffic, and load non-uniformity across output ports. Next, we considered buffer management strategies to support QoS functions, which utilize parameter values obtained via connection admission control (CAC) procedures to set tile threshold values. Through simulations, we showed that the buffer management schemes adopted behave well in the sense that they can protect regulated traffic from unregulated cell traffic in allocating buffer space. In particular, it was observed that dynamic partitioning is superior in terms of QoS support than virtual partitioning.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of KIISE:Computer Systems and Theory
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• v.30 no.10
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• pp.523-532
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• 2003

Video on Demand(VOD) services cause high resource consumption in a video seuer, because multimedia are characterized by continuous playback, a high bandwidth requirement, and long playback duration. Patching has been proposed to save the network I/O bandwidth of a video server. To achieve true VOD, patching uses multicasting to share video streams, thereby providing immediate VOD services to users without any service latency. A communication channel is used to either multicast the entire video as a regular channel or multicast only the leading portion of a video as a Patching channel. This paper Proposes a buffering patching technique that divides regular channels, as used in patching, into sub-regular channels and regular channels to shorten the holding time of the channels. In the proposed technique, the last portion of video data, corresponding to the size of the buffering window, is not transferred by sub-regular channels, but rather downloaded and buffered in the user buffer from the latest regular channel. When simulations were performed to compare the performance of the proposed technique with that of conventional patching, the results indicated that the proposed technique was superior in terms of the defection rate, average service latency, and fairness, although the amount of video data buffered at each user station was higher than that with patching.

• Journal of Broadcast Engineering
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• v.13 no.2
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• pp.179-187
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• 2008

In this paper, we proposed the qualify metric based on SVC and the subjective quality. The proposed quality metric is for a general purpose. It means we can use it for any video sequences regardless of its temporal and spatial characteristics. The Quality of Service(QoS) is one of the important issues in heterogeneous environment which has diverse restrictions such as limited network bandwidth and limited display resolution. Scalable Video Coding(SVC) is the efficient video coding skill in heterogeneous environment. Because SVC can be adapted to various quality bitstreams using three scalabilities(spatial, temporal, and SNR) from one bitstream which has full scalability. To maximize the QoS in this environment, we should consider the subjective quality which is the viewer response. And also we should consider temporal and spatial characteristics of video sequence because the subjective quality is affected by temporal and spatial characteristics of video sequence. To verify the efficiency of the proposed method, we perform subjective assessments. The experimental results show that the proposed method has high correlation with subjective quality. The proposed method can be a decision tool of SVC birstream extraction.

• Journal of Broadcast Engineering
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• v.13 no.2
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• pp.222-235
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• 2008

Recently, as growing of high definition media services like HDTV and IPTV, fast moving picture manipulation techniques need to meet what those services require. Especially, a fast reduced-size image extracting method is required in the areas of video indexing and video summary Conventional DC image extracting methods, however, can't be applied to H.264/AVC streams since a spatial domain prediction scheme is adopted in H.264/AVC intra mode. In this paper, we propose a theoretical method for extracting a thumbnail image from an H.264/AVC intra frame in the frequency domain. Furthermore, the proposed scheme can extract the thumbnail very fast since all operations are applied to transform coefficients directly, after a general equation for the thumbnail extraction in nine H.264/AVC intra prediction modes is introduced, an LUT(Look Up Table) for each mode is designed. Through the implementation and performance evaluation, while the subject quality difference between the output of our scheme and a conventional output is negligible, the former can extract the thumbnail faster then the latter by up to 63%.