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Journal articles on the topic 'Low-bitrate'

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Author: Grafiati

Published: 8 June 2024

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1

Rudzki, Tomasz, Ignacio Gomez-Lanzaco, Jessica Stubbs, Jan Skoglund, Damian T. Murphy, and Gavin Kearney. "Auditory Localization in Low-Bitrate Compressed Ambisonic Scenes." Applied Sciences 9, no. 13 (June 28, 2019): 2618. http://dx.doi.org/10.3390/app9132618.

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The increasing popularity of Ambisonics as a spatial audio format for streaming services poses new challenges to existing audio coding techniques. Immersive audio delivered to mobile devices requires an efficient bitrate compression that does not affect the spatial quality of the content. Good localizability of virtual sound sources is one of the key elements that must be preserved. This study was conducted to investigate the localization precision of virtual sound source presentations within Ambisonic scenes encoded with Opus low-bitrate compression at different bitrates and Ambisonic orders (1st, 3rd, and 5th). The test stimuli were reproduced over a 50-channel spherical loudspeaker configuration and binaurally using individually measured and generic Head-Related Transfer Functions (HRTFs). Participants were asked to adjust the position of a virtual acoustic pointer to match the position of virtual sound source within the bitrate-compressed Ambisonic scene. Results show that auditory localization in low-bitrate compressed Ambisonic scenes is not significantly affected by codec parameters. The key factors influencing localization are the rendering method and Ambisonic order truncation. This suggests that efficient perceptual coding might be successfully used for mobile spatial audio delivery.

2

van Schalkwyk, J. J. D. "Low bitrate video coding with depth compensation." IEE Proceedings - Vision, Image, and Signal Processing 141, no. 3 (1994): 149. http://dx.doi.org/10.1049/ip-vis:19941056.

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3

Wu, Dapeng, Linfeng Cui, Tong Tang, and Ruyan Wang. "Adaptive Bandwidth Prediction and Smoothing Glitches in Low-Latency Live Streaming." Security and Communication Networks 2022 (May 9, 2022): 1–13. http://dx.doi.org/10.1155/2022/4992957.

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HTTP adaptive streaming (HAS) technologies such as dynamic adaptive streaming over HTTP (DASH) and common media application format (CMAF) are now used extensively to deliver live streaming services to large numbers of viewers. However, in dynamic networks, inaccurate bandwidth prediction may result in the wrong request of bitrate, and short-term network fluctuations may produce glitches, causing unnecessary bitrate switching, thereby degrading clients' Quality of Experience (QoE). To tackle this, we propose adaptive bandwidth prediction and smoothing glitches in low-latency live streaming (called APSG) in this article. Concretely, firstly, the size of random bandwidth fluctuations is exploited as the weight of exponentially weighted moving average (EWMA) for adaptive bandwidth prediction; in addition to bandwidth prediction and buffer occupancy, glitches phenomena under a stable network environment are taken into account to enhance the viewing experience of clients. Finally, experimental results show that compared to traditional ABR algorithms under a stable network environment, APSG could reduce the number of bitrate switches and latency by up to 72.6% and 27.3%, respectively; under a dynamic network environment, APSG could reduce the number of bitrate switches and latency by up to 53.8% and 23.6%, respectively.

4

Kim, Hyung Yong, Ji Won Yoon, Won Ik Cho, and Nam Soo Kim. "Neurally Optimized Decoder for Low Bitrate Speech Codec." IEEE Signal Processing Letters 29 (2022): 244–48. http://dx.doi.org/10.1109/lsp.2021.3132557.

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5

Winkler, S., and C. Faller. "Perceived Audiovisual Quality of Low-Bitrate Multimedia Content." IEEE Transactions on Multimedia 8, no. 5 (October 2006): 973–80. http://dx.doi.org/10.1109/tmm.2006.879871.

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6

Georgis, Georgios, George Lentaris, and Dionysios Reisis. "Reduced Complexity Superresolution for Low-Bitrate Video Compression." IEEE Transactions on Circuits and Systems for Video Technology 26, no. 2 (February 2016): 332–45. http://dx.doi.org/10.1109/tcsvt.2015.2389431.

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7

Chandrasekhar, Vijay, Gabriel Takacs, David M. Chen, Sam S. Tsai, Yuriy Reznik, Radek Grzeszczuk, and Bernd Girod. "Compressed Histogram of Gradients: A Low-Bitrate Descriptor." International Journal of Computer Vision 96, no. 3 (May 15, 2011): 384–99. http://dx.doi.org/10.1007/s11263-011-0453-z.

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8

Ebrahimi, T., E. Reusens, and Wei Li. "New trends in very low bitrate video coding." Proceedings of the IEEE 83, no. 6 (June 1995): 877–91. http://dx.doi.org/10.1109/5.387090.

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9

Duan, Juzheng, Min Zhang, Jing Wang, Shuai Han, Xun Chen, and Xiaolong Yang. "VCC-DASH: A Video Content Complexity-Aware DASH Bitrate Adaptation Strategy." Electronics 9, no. 2 (January 31, 2020): 230. http://dx.doi.org/10.3390/electronics9020230.

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Traditional DASH (dynamic adaptation streaming over HTTP(i.e., HyperText Transfer Protocol)) bitrate strategy cannot differentiate segments with different complexities of video content, resulting in the user’s QoE (quality of experience) of segments with high content complexity as worse than that with low content complexity. In case of this, this paper firstly studies video coding and puts forward the definition of video content complexity. Then the effects of content complexity on user’s QoE is analyzed and the QoE utility function of the segment is formulated based on its MOS (mean opinion score, related to the content complexity and bitrate) and bitrate switching between consecutive segments. Last, in order to maximize user’s QoE, this paper proposes VCC-DASH (video content complexity-aware DASH bitrate adaptation strategy) under the constraints of the network bandwidth and the buffer occupancy. In simulations, we compare VCC-DASH with the classical bitrate adaptation strategy proposed by Liu et al. (LIU’s strategy, for short). The simulation results show that the two strategies have similar performances in bitrate switching numbers, playback interruption times, and buffer lengths. In addition, it is more important for simulation results to reveal that VCC-DASH’s average bitrate is much higher than that of LIU’s strategy, which means that VCC-DASH can make fuller use of the network bandwidth than LIU’s strategy does. Moreover, the MOS distribution of the VCC-DASH is more concentrated on the better scores “4~5”, which profit from its content complexity-aware adaptation to allocate more bandwidth resources to high-complexity segments.

10

Liang, Xiaoyun, Zhaohong Li, Zhonghao Li, and Zhenzhen Zhang. "Fake Bitrate Detection of HEVC Videos Based on Prediction Process." Symmetry 11, no. 7 (July 15, 2019): 918. http://dx.doi.org/10.3390/sym11070918.

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In order to defraud click-through rate, some merchants recompress the low-bitrate video to a high-bitrate one without improving the video quality. This behavior deceives viewers and wastes network resources. Therefore, a stable algorithm that detects fake bitrate videos is urgently needed. High-Efficiency Video Coding (HEVC) is a worldwide popular video coding standard. Hence, in this paper, a robust algorithm is proposed to detect HEVC fake bitrate videos. Firstly, five effective feature sets are extracted from the prediction process of HEVC, including Coding Unit of I-picture/P-picture partitioning modes, Prediction Unit of I-picture/P-picture partitioning modes, Intra Prediction Modes of I-picture. Secondly, feature concatenation is adopted to enhance the expressiveness and improve the effectiveness of the features. Finally, five single feature sets and three concatenate feature sets are separately sent to the support vector machine for modeling and testing. The performance of the proposed algorithm is compared with state-of-the-art algorithms on HEVC videos of various resolutions and fake bitrates. The results show that the proposed algorithm can not only can better detect HEVC fake bitrate videos, but also has strong robustness against frame deletion, copy-paste, and shifted Group of Picture structure attacks.

11

Li, Mao-quan, and Zheng-quan Xu. "An adaptive preprocessing algorithm for low bitrate video coding." Journal of Zhejiang University-SCIENCE A 7, no. 12 (December 2006): 2057–62. http://dx.doi.org/10.1631/jzus.2006.a2057.

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12

Garcia, Ray, Velibor Adzic, and Hari Kalva. "Adapting Low-Bitrate Skip Mode in a Mobile Environment." IEEE Transactions on Circuits and Systems for Video Technology 27, no. 2 (February 2017): 352–65. http://dx.doi.org/10.1109/tcsvt.2014.2365653.

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13

Tang, Tong, and Ling Li. "Adaptive deblocking method for low bitrate coded HEVC video." Journal of Visual Communication and Image Representation 38 (July 2016): 721–34. http://dx.doi.org/10.1016/j.jvcir.2016.04.002.

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14

Rijkse, Karel. "ITU standardisation of very low bitrate video coding algorithms." Signal Processing: Image Communication 7, no. 4-6 (November 1995): 553–65. http://dx.doi.org/10.1016/0923-5965(95)00023-3.

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15

Huang, Hao, Li Guo, and Lin Li. "Audio Watermarking for MP3 Coding with Low Bitrate and Low Sampling Frequency." Journal of Electronics & Information Technology 30, no. 11 (April 15, 2011): 2678–83. http://dx.doi.org/10.3724/sp.j.1146.2007.00666.

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16

Xu, Ke, Bin Liu, Yongjian Nian, Mi He, and Jianwei Wan. "Distributed lossy compression for hyperspectral images based on multilevel coset codes." International Journal of Wavelets, Multiresolution and Information Processing 15, no. 02 (February 7, 2017): 1750012. http://dx.doi.org/10.1142/s0219691317500126.

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This paper focuses on the problem of lossy compression for hyperspectral images and presents an efficient compression algorithm based on distributed source coding. The proposed algorithm employs a block-based quantizer followed by distributed lossless coding, which is implemented through the use of multilevel coset codes. First, a bitrate allocation algorithm is proposed to assign the rational bitrate for each block. Subsequently, the multilinear regression model is employed to construct the side information of each block, and the optimal quantization step size of each block is obtained under the assigned bitrate while minimizing the distortion. Finally, the quantized version of each block is encoded by distributed lossless compression. Experimental results show that the compression performance of the proposed algorithm is competitive with that of state-of-the-art transform-based compression algorithms. Moreover, the proposed algorithm provides both low encoder complexity and error resilience, making it suitable for onboard compression.

17

Andruloniw, Paweł, Karol Kowalik, and Piotr Zwierzykowski. "Unsupervised Learning Data-Driven Continuous QoE Assessment in Adaptive Streaming-Based Television System." Applied Sciences 12, no. 16 (August 19, 2022): 8288. http://dx.doi.org/10.3390/app12168288.

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The quality of experience (QoE) assessment of adaptive video streaming may be crucial for detecting degradations impacting customer satisfaction. In a telecommunication environment, eliminating failure points may be the highest priority. This study aims to assess the QoE level of the video played by the STB device connected to the production TV system. The evaluation has been based on the stalling effects, video quality changes, and the time related to the last decreased bitrate change occurrence. The two-phase continuous clustering approach has been studied to assess the QoE level based on the ACR scale. The number of devices with grades 1 or 2 is relatively low, but those devices generate significantly more events than adequately functioning devices. STBs try to play the highest possible bitrate, and there is no possibility of setting the intermediate bitrate level. The STB player does not have the button to set the quality level, usually available in pure over-the-top applications. Hence the bitrate fluctuations that can annoy customers appear for the lowest grades. The boundary cases can be easily assessed. The outcome should be challenged by the customers’ opinions to find the proper QoE threshold. Continuous clustering may allow telecom operators to assess customer satisfaction with their TV service.

18

Hines, Andrew, Eoin Gillen, Damien Kelly, Jan Skoglund, Anil Kokaram, and Naomi Harte. "ViSQOLAudio: An objective audio quality metric for low bitrate codecs." Journal of the Acoustical Society of America 137, no. 6 (June 2015): EL449—EL455. http://dx.doi.org/10.1121/1.4921674.

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19

Abdul Rahman, Farah Diyana, Dimitris Agrafiotis, and Ahmad Imran Ibrahim. "Edge Dissimilarity Reduced-Reference Quality Metric with Low Overhead Bitrate." Indonesian Journal of Electrical Engineering and Computer Science 10, no. 2 (May 1, 2018): 631. http://dx.doi.org/10.11591/ijeecs.v10.i2.pp631-640.

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In multimedia transmission, it is important to rely on an objective quality metric which accurately represents the subjective quality of processed images and video sequences. Reduced-reference metrics make use of side-information that is transmitted to the receiver for estimating the quality of the received sequence with low complexity. In this paper, an Edge-based Dissimilarity Reduced-Reference video quality metric with low overhead bitrate is proposed. The metric is evaluated by finding the dissimilarity between the edge information of original and distorted sequences. The edge degradation can be detected in this manner as perceived video quality is highly associated with edge structural. Due to the high overhead using the Soergel distance, it is pertinent to find a way to reduce the overhead while maintaining the edge information that can convey the quality measure of the sequences. The effects of different edge detection operator, video resolution and file compressor are investigated. The aim of this paper is to significantly reduce the bitrate required in order to transmit the side information overhead as the reduced reference video quality metric. From the results obtained, the side information extracted using Sobel edge detector maintained consistency throughout the reduction of spatial and temporal down-sample.

20

Mitrica, Iulia, Eric Mercier, Christophe Ruellan, Attilio Fiandrotti, Marco Cagnazzo, and Beatrice Pesquet-Popescu. "Very Low Bitrate Semantic Compression of Airplane Cockpit Screen Content." IEEE Transactions on Multimedia 21, no. 9 (September 2019): 2157–70. http://dx.doi.org/10.1109/tmm.2019.2900168.

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21

Herre, Jürgen, and Sascha Dick. "Psychoacoustic Models for Perceptual Audio Coding—A Tutorial Review." Applied Sciences 9, no. 14 (July 17, 2019): 2854. http://dx.doi.org/10.3390/app9142854.

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Psychoacoustic models of human auditory perception have found an important application in the realm of perceptual audio coding, where exploiting the limitations of perception and removal of irrelevance is key to achieving a significant reduction in bitrate while preserving subjective audio quality. To this end, psychoacoustic models do not need to be perfect to satisfy their purpose, and in fact the commonly employed models only represent a small subset of the known properties and abilities of the human auditory system. This paper provides a tutorial introduction of the most commonly used psychoacoustic models for low bitrate perceptual audio coding.

22

Jin, Liu, and Yiwen Zhang. "An Improved NID Steganography Method for Low Bitrate Speech on VoIP." Modern Electronic Technology 5, no. 1 (May 6, 2021): 1. http://dx.doi.org/10.26549/met.v5i1.6167.

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In the procedure of encoding process on low bitrate speech, fixed codebook division is an efficient and promising embedding method for steganography. An improved neighbor index division (NID) steganography method based on the high bitrate frame of G.723.1 codec (6.3kbit/s) is proposed, which employs the parity and low distortion of neighbor indices for G.723.1 fixed codebooks. Differing from previously NID method which performs quantized index modulation (QIM) beforehand, the proposed method divides codeword indices into separate sub-codebooks according to the secret message bits dynamically in the original G.723.1 codec quantization period. Compared with existing NID method, our proposed method doesn’t need to divide the codebook before the encoding starts. The embedding and codebook dividing happen simultaneously, which utilizes the characteristics of specific secret message bits. The experiment results show that the proposed method has a much lower quality degradation for the decoding speech and still fulfills the low latency requirement for communication.

23

Jin Wang, Peilin Liu, Ji Kong, and Rendong Ying. "Split Table Extension: A Low Complexity LVQ Extension Scheme in Low Bitrate Audio Coding." IEEE Signal Processing Letters 17, no. 1 (January 2010): 59–62. http://dx.doi.org/10.1109/lsp.2009.2032982.

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24

Edan, Noor N., and Nasser N. Khamiss. "HIGH QUALITY LOW BITRATE VOICE CODEC FOR TRANSMISSION OVER ADVANCED LTE." Iraqi Journal of Information and Communications Technology 1, no. 1 (December 15, 2021): 83–93. http://dx.doi.org/10.31987/ijict.1.1.167.

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In mobile communication systems bit-rate reductions while maintaining an acceptable voice quality are necessary to achieve efficiency in channel bandwidth utilization and users satisfaction. As Long-Term Evolution(LTE) converging towards all-IP solutions and supporting VOIP service, the voice signals are converted into coded digital bit-stream and sent over the network. This paper proposes the implementation of codebook excited linear prediction (CELP) voice codec algorithm based on two source-rates of low 9.6Kbps and medium 16Kbps for achieving a perceptible level of voice quality, while efficiently using available bandwidth during the transmission over advanced LTE. The architecture of proposed CELP codec model is implemented to decompose the voice signal into a set of parameters that characterize each particular frame at the encoder part, these parameters are quantized and encoded for transmission to the decoder. The investigation showed that the configuration of the link and the applied CELP codec mode mainly influence on the obtained voice capacity and quality. The quantifying also shows that the voice quality can be traded for the enhanced capacity, since the low rate codec will produce lower voice quality than higher rate codec. Also, this paper is achieved, during theconfiguration of the system with higher channel quality indicator (CQI) index, increasing in the capacity gain to a saturated value of about 500 and 1000 users per cell over 5MHz bandwidth for transmit diversity (TD) and Open-Loop Spatial Multiplexing (OLSM) respectively and up to 1000 and 2000 users per cell over 10MHz channel bandwidth for TD and OLSM respectively.

25

Suseela, G., and Asnathvicty Phamila Y. "LOW BITRATE HYBRID SECURED IMAGE COMPRESSION FOR WIRELESS IMAGE SENSOR NETWORK." Asian Journal of Pharmaceutical and Clinical Research 10, no. 13 (April 1, 2017): 101. http://dx.doi.org/10.22159/ajpcr.2017.v10s1.19578.

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Wireless image sensor networks are capable of sensing, processing and transmitting the visual data along with the scalar data and have attainedwide attention in sensitive applications such as visual surveillance, habitat monitoring, and ubiquitous computing. The sensor nodes in the network are resource constrained in nature. Since the image data are huge always high computational cost and energy budget are levied on the sensor nodes. The compression standards JPEG and JPEG 2000 are not feasible as they involve complex computations. To stretch out the life span of these nodes,it is required to have low complex and low bitrate image compression techniques exclusively designed for this platform. The complicated scenarioof wireless sensor network in processing and transmitting image data has been addressed by a low complex hybrid secured image compression technique using discrete wavelet transform and Bin discrete cosine transformation.

26

Tao, Xiaoming, Linhao Dong, Shaoyang Li, Yang Li, Ning Ge, and Jianhua Lu. "Improving low bitrate video coding via computation incorporating a priori information." China Communications 12, no. 9 (2015): 22–34. http://dx.doi.org/10.1109/cc.2015.7275256.

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27

Bernard, A., and A. Alwan. "Low-bitrate distributed speech recognition for packet-based and wireless communication." IEEE Transactions on Speech and Audio Processing 10, no. 8 (November 2002): 570–79. http://dx.doi.org/10.1109/tsa.2002.808141.

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28

Backstrom, Tom, and Johannes Fischer. "Fast Randomization for Distributed Low-Bitrate Coding of Speech and Audio." IEEE/ACM Transactions on Audio, Speech, and Language Processing 26, no. 1 (January 2018): 19–30. http://dx.doi.org/10.1109/taslp.2017.2757601.

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29

Hasegawa, Keisuke, Kenichi Noguchi, Toshiharu Morizumi, Junichi Nakajima, Kouichi Furukado, Masakatsu Aoki, Tetsuya Yamaguchi, et al. "Evaluation of Low-Delay Audio Codec Appliance for Bitrate-limited Network." Journal of The Institute of Image Information and Television Engineers 74, no. 2 (2020): 395–401. http://dx.doi.org/10.3169/itej.74.395.

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30

Mallik, Bruhanth, Akbar Sheikh-Akbari, and Ah-Lian Kor. "Mixed-resolution HEVC based multiview video codec for low bitrate transmission." Multimedia Tools and Applications 78, no. 6 (July 28, 2018): 6701–20. http://dx.doi.org/10.1007/s11042-018-6272-2.

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31

Urata, Yuichiro, Masanori Koike, Kazuhisa Yamagishi, Noritsugu Egi, and Jun Okamoto. "Extension of ITU-T P.1203 model to Tile-based Omnidirectional Video Streaming." Electronic Imaging 2021, no. 11 (January 18, 2021): 161–1. http://dx.doi.org/10.2352/issn.2470-1173.2021.11.hvei-161.

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Omnidirectional video (ODV) streaming has become widespread. Since the data size of ODV is extremely large, tilebased streaming has been developed to compress the data. In this coding technology, high-quality tiles encoded at a higher bitrate for the users’ viewing direction and low-quality tiles encoded at a lower bitrate for the whole environment are sent to the client, and a player decodes these tiles. As a result, quality degrades due to coding, streaming, and the client’s buffer. Therefore, to provide high-quality tile-based ODV streaming services, qualityof- experience needs to be monitored by comprehensively evaluating the quality degradations. By taking into account the quality degradation due to the low-quality tiles, the ITU-T Recommendation P.1203 model, which can be used for monitoring the quality of 2D video streaming services, is extended to tile-based ODV streaming services. Our model is demonstrated to estimate quality with sufficiently high accuracy.

32

Liu, Peng, Jongwon Yoon, Ha Ryung Kim, and Suman Banerjee. "VideoCoreCluster: Energy-Efficient, Low-Cost, and Hardware-Assisted Video Transcoding System." Wireless Communications and Mobile Computing 2018 (June 5, 2018): 1–13. http://dx.doi.org/10.1155/2018/7470234.

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Video streaming is one of the killer applications in recent years. Video transcoding plays an important role in the video streaming service to cope with the various purposes. Specifically, content owners and publishers heavily utilize video transcoders to reconfigure source video in a variety of formats, video qualities, and bitrate to provide end users with the best possible quality of service. In this paper, we present VideoCoreCluster, a low-cost and energy-efficient transcoder cluster that is suitable for live streaming services. We designed and implemented real-time video transcoder cluster using cheap ($35), powerful, and energy-efficient Raspberry Pi. The quality of transcoded video provided by VideoCoreCluster is similar to the best software-based video transcoder while consuming significantly less energy (<3 W). We have proposed a scheduling algorithm based on priority of video stream and transcoding capacity. Our cluster manager provides reliable and scalable streaming services, because it uses the characteristics of adaptive bitrate scheme. We have deployed our transcoding cluster to provide IP-based TV streaming services on our university campus.

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Gusiev, O. Yu, V. І. Мagro, and O. I. Nikolska. "TELETRAFFIC FORECASTING IN MEDIA SERVICE SYSTEMS." Radio Electronics, Computer Science, Control, no. 4 (December 22, 2023): 7. http://dx.doi.org/10.15588/1607-3274-2023-4-1.

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Context. The development of information and communication technologies has led to an increase in the volume of information sent over the network. Media service platforms play an important role in the creation and processing of bitrate in the information network. Therefore, there is a need to develop a methodology for predicting bitrate in various media service platforms by creating an effective algorithm that minimizes the forecast error. Objective. The aim of the work is to synthesize in analytical form the state transition matrix of the Kalman filter for nonstationary self-similar processes when predicting the bitrate in telecommunication networks. Method. A methodology has been developed for predicting teletraffic in media service platforms, based on a modification of the Kalman filter for non-Gaussian processes. This methodology uses an original procedure for calculating statistics, which makes it possible to reduce the filtering and forecast error that arises due to the uncertainty of the analytical model of the process under study. The methodology does not require knowledge of the analytical model of the process, as well as strict restrictions on its stochastic characteristics. Results. A methodology for estimating and forecasting bitrate in telecommunication systems is proposed. This methodology was used to study teletraffic processes in the media service platforms Google Meet, Zoom, Microsoft Teams. The passage of real bitrate through the specified media service platforms was studied. A comparison of real teletraffic with predicted teletraffic was carried out. The influence of the order of the state transition matrix of the Kalman filter on the error of estimation and prediction has been studied. It has been established that even a low (second) order of the state transition matrix allows one to obtain satisfactory forecast results. It is shown that the use of the proposed methodology makes it possible to predict traffic with a relative error of the order of 3– 4%. Conclusions. An original algorithm for assessing and forecasting the characteristics of media traffic has been developed. Recommendations for improving the technology for building media service platforms are formulated. It is shown that the bitrates generated by various media service platforms, in the case of applying the proposed estimation and forecasting methodology, are invariant with respect to the type of stochastic processes being processed.

34

Tang, Tong, Ling Li, Xiaoyu Wu, Ruizhi Chen, Haochen Li, Guo Lu, and Limin Cheng. "TSA-SCC: Text Semantic-Aware Screen Content Coding With Ultra Low Bitrate." IEEE Transactions on Image Processing 31 (2022): 2463–77. http://dx.doi.org/10.1109/tip.2022.3152003.

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35

Wang, Ziyi, Yong Cui, Xiaoyu Hu, Xin Wang, Wei Tsang Ooi, Zhen Cao, and Yi Li. "MultiLive: Adaptive Bitrate Control for Low-Delay Multi-Party Interactive Live Streaming." IEEE/ACM Transactions on Networking 30, no. 2 (April 2022): 923–38. http://dx.doi.org/10.1109/tnet.2021.3129481.

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36

Andrei, Silviu S., Nataliya Shapovalova, and Walterio Mayol-Cuevas. "SUPERVEGAN: Super Resolution Video Enhancement GAN for Perceptually Improving Low Bitrate Streams." IEEE Access 9 (2021): 91160–74. http://dx.doi.org/10.1109/access.2021.3090344.

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37

Lee, J. S., J. W. Kim, K. H. Lee, and S. J. Ko. "Residual motion coding method supporting error concealment in low bitrate video coding." Electronics Letters 41, no. 7 (2005): 406. http://dx.doi.org/10.1049/el:20057737.

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38

Schiller, H., and B. B. Chaudhuri. "Efficient coding of side information in a low bitrate hybrid image coder." Signal Processing 19, no. 1 (January 1990): 61–73. http://dx.doi.org/10.1016/0165-1684(90)90007-l.

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Mouchtaris, Athanasios, Kiki Karadimou, and Panagiotis Tsakalides. "Multiresolution Source/Filter Model for Low Bitrate Coding of Spot Microphone Signals." EURASIP Journal on Audio, Speech, and Music Processing 2008 (2008): 1–16. http://dx.doi.org/10.1155/2008/624321.

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40

De Natale, Francesco G. B., and Daniele D. Giusto. "A mesh-interpolation scheme for very-low bitrate coding of video sequences." European Transactions on Telecommunications 9, no. 1 (January 1998): 47–55. http://dx.doi.org/10.1002/ett.4460090106.

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Tsuji, Hiroyuki, Shinji Tokumasu, Hiroki Takahashi, and Masayuki Nakajima. "Spatial prefiltering scheme based on anisotropic diffusion in low-bitrate video coding." Systems and Computers in Japan 38, no. 10 (2007): 34–45. http://dx.doi.org/10.1002/scj.20566.

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Deng, Jun, Björn Schuller, Florian Eyben, Dagmar Schuller, Zixing Zhang, Holly Francois, and Eunmi Oh. "Exploiting time-frequency patterns with LSTM-RNNs for low-bitrate audio restoration." Neural Computing and Applications 32, no. 4 (May 10, 2019): 1095–107. http://dx.doi.org/10.1007/s00521-019-04158-0.

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Brown, Harrison, Kai Fricke, and Eiko Yoneki. "World-Models for Bitrate Streaming." Applied Sciences 10, no. 19 (September 24, 2020): 6685. http://dx.doi.org/10.3390/app10196685.

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Abstract:

Adaptive bitrate (ABR) algorithms optimize the quality of streaming experiences for users in client-side video players, especially in unreliable or slow mobile networks. Several rule-based heuristic algorithms can achieve stable performance, but they sometimes fail to properly adapt to changing network conditions. Fluctuating bandwidth may cause algorithms to default to behavior that creates a negative experience for the user. ABR algorithms can be generated with reinforcement learning, a decision-making paradigm in which an agent learns to make optimal choices through interactions with an environment. Training reinforcement learning algorithms for bitrate streaming requires building a simulator for an agent to experience interactions quickly; training an agent in the real environment is infeasible due to the long step times in real environments. This project explores using supervised learning to construct a world-model, or a learned simulator, from recorded interactions. A reinforcement learning agent that is trained inside of the learned model, rather than a simulator, can outperform rule-based heuristics. Furthermore, agents that are trained inside the learned world-model can outperform model-free agents in low sample regimes. This work highlights the potential for world-models to quickly learn simulators, and to be used for generating optimal policies.

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Bouaafia, Soulef, Randa Khemiri, Seifeddine Messaoud, and Fatma Elzahra Sayadi. "Complexity Analysis of New Future Video Coding (FVC) Standard Technology." International Journal of Digital Multimedia Broadcasting 2021 (August 2, 2021): 1–13. http://dx.doi.org/10.1155/2021/6627673.

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Future Video Coding (FVC) is a modern standard in the field of video coding that offers much higher compression efficiency than the HEVC standard. FVC was developed by the Joint Video Exploration Team (JVET), formed through collaboration between the ISO/IEC MPEG and ITU-T VCEG. New tools emerging with the FVC bring in super resolution implementation schemes that are being recommended for Ultra-High-Definition (UHD) video coding in both SDR and HDR images. However, a new flexible block structure is adopted in the FVC standard, which is named quadtree plus binary tree (QTBT) in order to enhance compression efficiency. In this paper, we provide a fast FVC algorithm to achieve better performance and to reduce encoding complexity. First, we evaluate the FVC profiles under All Intra, Low-Delay P, and Random Access to determine which coding components consume the most time. Second, a fast FVC mode decision is proposed to reduce encoding computational complexity. Then, a comparison between three configurations, namely, Random Access, Low-Delay B, and Low-Delay P, is proposed, in terms of Bitrate, PSNR, and encoding time. Compared to previous works, the experimental results prove that the time saving reaches 13% with a decrease in the Bitrate of about 0.6% and in the PSNR of 0.01 to 0.2 dB.

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Zhang, Jialiang, Guanjun Gao, Jingwen Li, Ziqi Ma, and Yonggang Guo. "Experimental Demonstration and Simulation of Bandwidth-Limited Underwater Wireless Optical Communication with MLSE." Photonics 9, no. 3 (March 12, 2022): 182. http://dx.doi.org/10.3390/photonics9030182.

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Underwater wireless optical communication (UWOC) is able to provide large bandwidth, low latency, and high security. However, there still exist bandwidth limitations in UWOC systems, with a lack of effective compensation methods. In this paper, we systematically study the bandwidth limitation due to the transceiver and underwater channel through experiments and simulations, respectively. Experimental results show that by using the 7-tap maximum likelihood sequence estimation (MLSE) detection, the maximum bitrate of the simple rectangular shape on–off-keying (OOK) signaling is increased from 2.4 Gb/s to 4 Gb/s over 1 GHz transceiver bandwidth, compared to the conventional symbol-by-symbol detection. For the bandwidth limitation caused by the underwater channel, we simulate the temporal dispersion in the UWOC by adopting a Monte Carlo method with a Fournier–Forand phase function. With MLSE adopted at the receiver, the maximum available bitrate is improved from 0.4 to 0.8 Gb/s in 12 m of harbor water at the threshold of hard-decision forward-error-correction (HD-FEC, 3.8 × 10−3). Moreover, when the bitrate for 0.4 Gb/s 12 m and 0.8 Gb/s 10 m OOK transmission remains unchanged, the power budget can be reduced from 33.8 dBm to 30 dBm and from 27.8 dBm to 23.6 dBm, respectively. The results of both experiments and simulations indicate that MLSE has great potential for improving the performance of bandwidth-limited communication systems.

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Gandam, Anudeep, and Jagroop Singh Sidhu. "Fuzzy Based Adaptive Deblocking Filters at Low-Bitrate HEVC Videos for Communication Networks." Computers, Materials & Continua 66, no. 3 (2021): 3045–63. http://dx.doi.org/10.32604/cmc.2021.013663.

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Korobeynikov, A. V., M. A. Boyarshinov, A. I. Nistyuk, and V. N. Emelianov. "Using the Standard P.862 to Compare the Quality of Low-Bitrate Vocoders." Intellekt. Sist. Proizv. 16, no. 4 (February 25, 2019): 109. http://dx.doi.org/10.22213/2410-9304-2018-4-109-113.

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Рассматриваются объективные методы оценки качества речевого сигнала: 1) Perceptual Evaluation of Speech (PESQ, рекомендация МСЭ-T P.862) - оценка восприятия качества речи, 2) Listening Quality Objective (LQO, рекомендация, МСЭ-T P.800.1) - качество прослушивания. Приведено краткое описание и схемы работы методики PESQ и формулы для преобразования оценок Raw MOS в MOS-LQO и обратно. Для тестирования были выбраны низкоскоростные вокодеры: 1) MEPLe, 2) Speex, 3) Codec2. Тестирование вокодеров проводилось на битовых скоростях от 700 до 4800 бит/с. Для тестирования использовались аудиофайлы артикуляционных таблиц количеством 20 записей (wav, 8000 КГц, 16 бит, моно). В результате тестирования были построены таблицы и графики для Raw MOS и MOS-LQO оценок выбранных вокодеров. При анализе результатов экспериментов сделан вывод об эффективности применения объективных методов оценки качества речи, и в качестве перспективного для дальнейших разработок вокодера был выделен MELPe, обеспечивающий на битовых скоростях 1200 и 2400 бит/с оценку качества MOS соответственно 2,9...3,2 и 3,0...3,3. Вокодер Speex показал сравнимые с MELPe результаты оценки при большей битовой скорости (4800 бит/с), а вокодер Codec2 показал результаты оценки ниже, чем MELPe.

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Habchi, Yassine, Ameur Fethi Aimer, Mohammed Beladgham, and Riyadh Bouddou. "Ultra low bitrate retinal image compression using integer lifting scheme and subband encoder." Indonesian Journal of Electrical Engineering and Computer Science 24, no. 1 (October 1, 2021): 295. http://dx.doi.org/10.11591/ijeecs.v24.i1.pp295-307.

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Recently, ophthalmic clinics have seen many complaints related to retinal diseases. The degree of clarity of the blood vessels (BV) in the eye can be an important indicator of some diseases affecting the retina such as diabetic retinopathy. To diagnose it, we need to intervene more than a medical team, especially in some difficult cases, through the exchange of medical images obtained by photography. This method has contributed significantly to the production of large data that can quickly saturate transmission, storage systems and increase processing time, so the need to compress images efficiently without modifying the content before transmission represents a major challenge. This paper provides an effective method for compressing color retinal images (CRI), which relies on the use of an integer lifting scheme (ILS) based on cohen daubechies-feaveau wavelet (CDFW9/7) and the set partitioning in hierarchical trees (SPIHT) to encode large coefficients. The obtained results demonstrate that the suggested method reduce algorithmic complexity, improve the retinal image quality and achieves high objective parameters values for ultra-low bitrate compared to the conventional methods.

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Vijayanagar, K. R., and J. Kim. "Real-time low-bitrate multimedia communication for smart spaces and wireless sensor networks." IET Communications 5, no. 17 (November 25, 2011): 2482–90. http://dx.doi.org/10.1049/iet-com.2010.0929.

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Imura, Koji, and Yutaka Machida. "Error resilient video coding schemes for real-time and low-bitrate mobile communications." Signal Processing: Image Communication 14, no. 6-8 (May 1999): 519–30. http://dx.doi.org/10.1016/s0923-5965(99)00003-x.

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