• Journal of Broadcast Engineering
• /
• v.27 no.4
• /
• pp.548-560
• /
• 2022

Video based Point Cloud Compression (V-PCC) is one of the compression methods for compressing point clouds, and shows high efficiency in dynamic point cloud compression with movement due to the feature of compressing point cloud data using an existing video codec. Accordingly, V-PCC is drawing attention as a core technology for immersive content services such as AR/VR. In order to effectively service these V-PCC contents through a media streaming platform, it is necessary to encapsulate them in the existing media file format, ISO based Media File Format (ISOBMFF). However, in order to service through an adaptive streaming platform such as Dynamic Adaptive Streaming over HTTP (DASH), it is necessary to encode V-PCC contents of various qualities and store them in the server. Due to the size of the 2D media, it causes a great burden on the encoder and the server compared to the existing 2D media. As a method to solve such a problem, it may be considered to configure a streaming platform based on content obtained through V-PCC content encoding based on SHVC. Therefore, this paper encapsulates the SHVC-based V-PCC bitstream into ISOBMFF suitable for DASH service and proposes a configuration method to service it. In addition, in this paper, we propose ISOBMFF encapsulation and DASH configuration method to effectively service SHVC-based V-PCC contents, and confirm them through verification experiments.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.4
• /
• pp.392-400
• /
• 2009

In this paper, we propose an effective speech reinforcement technique base on ITU-T G.729A CS-ACELP codec under the near-end background noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. In contrast to the conventional speech reinforcement algorithm, we reinforce the excitation signal of the codec's parameters received from the far-end speech signal based on the G.729A speech codec under various background noise environments. Specifically, we first estimate the excitation signal of ambient noise at the near-end through the encoder of the G.729A speech codec, reinforcing the excitation signal of the far-end speech transmitted from the far-end. we specially propose a novel approach to directly reinforce the excitation signal of far-end speech signal based on the decoder of the G.729A. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with conventional SNR Recovery methods.

• Journal of the Korea Institute of Information Security & Cryptology
• /
• v.24 no.1
• /
• pp.51-58
• /
• 2014

A LFSR is commonly used for various stream cryptography applications to generate random numbers. A Leap-ahead LFSR was presented to generate a multi-bits random number per cycle. It only requires a single LFSR and it has an advantages in hardware complexity. However, it suffers from the significant reduction of maximum period of the generated random numbers. This paper presents the new segmented Leap-ahead LFSR to solve this problem. It consists of two segmented LFSRs. We prove the efficiency of the proposed segmented architecture using the precise mathematical analysis. We also demonstrate the proposed comparison results with other counterparts using Xinilx Vertex5 FPGA. The proposed architecture can increase 2.5 times of the maximum period of generated random numbers compared to the typical Leap-ahead architecture.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2014.06a
• /
• pp.258-261
• /
• 2014

The era of realistic broadcast with high fidelity has come after the wide-spread distribution of UHD display and the transmission of UHD experimental broadcast in CATV. However, UHD broadcast now has constraint because it requires much amount of bandwidth and data in broadcasting transmission and production system. Not only HEVC(High Efficiency Video Codec) which has more than two times higher compression rate but also cloud-based editing system would be the key to solve the problems above. Also, fast scene change detection of videos is needed to index and search UHD videos smoothly. In this paper, therefore, a method is proposed to index and search the scene change information of large volume UHD videos compressed with high-efficiency codec. Application usages of fast detection of scene change information in various UHD video environments are considered by using this algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.7C
• /
• pp.654-662
• /
• 2009

Video streams for mobile video streaming can be encoded to fit the available network bandwidth by controlling three factors: temporal resolution, spatial resolution, and picture quality. The controlling of picture quality by modifying the quantization parameter (QP) is most widely used. In this paper, we demonstrate that reducing the spatial resolution adaptively can be more efficient in terms of picture quality and energy consumption in low bit-rate environment, and present a model to find the optimal spatial resolution for the available bandwidth. Adaptive spatial resolution control scheme is especially effective when the bandwidth between the video server and the mobile device varies considerably with time, and when the mobile device is sensitive to energy consumption. Our scheme can improve the picture quality by approximately O.5dB and reduce energy consumption by more than 50% compared to the conventional video coding in low bit-rate environment.

• Journal of Broadcast Engineering
• /
• v.14 no.4
• /
• pp.387-396
• /
• 2009

No-Reference video-quality assessments are divided into two kinds of metrics based on decoding pixel domain or the bitstream one. Traditional full-/reduced- reference methods have difficulty to be deployed as realtime video transmission because it has problems of additional data, complexity, and assessment accuracy. This paper presents simple and highly accurate no-reference video-quality assessment in realtime video transmission. Our proposed method uses quantization parameter, motion vector, and information of transmission error. To evaluate performance of the proposed algorithm, we perform subjective test of video quality with the ITU-T P.910 Absolute Category Rating(ACR) method and compare our proposed algorithm with the subjective quality assessment method. Experimental results show the proposed quality metric has a high correlation (85%) in terms of subjective quality assessment.

• Journal of Broadcast Engineering
• /
• v.6 no.2
• /
• pp.121-130
• /
• 2001

We propose an efficient DCT-domain video transcoding algorithm for flexible for bit-rate video communications. Video transcoding provides communication fiexibility by adaptively changing the bit-rate of compressed bit stream. During the transcoding process, adrift error is unavoidable because of the difference between reference images in the series of encoding and decoding. For solving the drift error problem, cascade pixel-domain transcoder (CPDT) has been proposed. CPDT, however, requires highly complex hardware and heavy computational overhead. In this paper we propose a DCT-domain transcoding technique, which enables efficient transcoding without any drift error. The proposed cascade DCT-domain transcoder (CDDT) is realized by new motion compensation and down-sampling methods in the DCT-domain.

• The Journal of the Korea Contents Association
• /
• v.12 no.4
• /
• pp.10-19
• /
• 2012

Seamless streaming of multimedia content that ensures Quality of Service over heterogeneous networks has been a desire for many multimedia services, for which the multimedia contents should be adapted to usage environments such as network characteristics, terminal capabilities, and user preferences. Scalability in video coding is a good feature to meet the requirement of heterogeneous networks. In this paper, we propose a dynamic adaptation scheme of H.264/AVC SVC bit-stream using the MPEG-21 Digital Item Adaptation (DIA) tool. MPEG-21 DIA framework provides systematic solutions in choosing an adaptation operation to given conditions and supports interoperable video adaptation. The experiment results show that the proposed adaptation scheme provides QoS-enabled delivery and consumption of SVC with time-varying constraints of network, terminal, and user preference, in a robust and efficient way. In particular, the proposed adaptation scheme is proved to work well with very low delay under the condition that the variation rate of the given network bandwidth is upto 62%.

• Journal of Broadcast Engineering
• /
• v.14 no.3
• /
• pp.332-341
• /
• 2009

Terrestrial digital multimedia broadcasting(T-DMB) is one of mobile broadcasting services, and the commercial service was started in December 2005 in Korea. The performance targets of T-DMB are providing VCD(video CD) quality video and FM radio quality audio. In recent years, the researches for providing high quality video or audio service on T-DMB environments have been being carried out. To provide high-quality video or audio service, some additional data should be transmitted to the receiver as well as T-DMB video and audio data. Since the data rate for one T-DMB program is low, it is important to transmit the additional data at a low bit rate. In this paper, we propose a packetizing method for efficient transmission of the additional data to provide multichannel audio service on T-DMB environment.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.14 no.1
• /
• pp.197-204
• /
• 2010

This paper proposes a new body-effect compensated switch configuration for low voltage and low distortion switched-capacitor (SC) applications. The proposed circuit allows rail-to-rail switching operation for low voltage SC circuits and has better total harmonic distortion than the conventional bootstrapped circuit by 19 dB. A 2-1 cascaded sigma-delta modulator is provided for performing the high-resolution analog-to-digital conversion on audio codec in a communication transceiver. An experimental prototype for a single-stage folded-cascode operational amplifier (opamp) and a 2-1 cascaded sigma-delta modulator has been implemented m a 0.25 micron double-poly, triple-metal standard CMOS process with 2.7 V of supply voltage. The 1% settling time of the opamp is measured to be 560 ns with load capacitance of 16 pF. The experimental testing of the sigma-delta modulator with bit-stream inspection and analog spectrum analyzing plot is performed. The die size is $1.9{\times}1.5\;mm$.