Journal articles on the topic 'Codes for low-latency streaming'

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1

Nikhil Krishnan, M., Vinayak Ramkumar, Myna Vajha, and P. Vijay Kumar. "Simple Streaming Codes for Reliable, Low-Latency Communication." IEEE Communications Letters 24, no. 2 (February 2020): 249–53. http://dx.doi.org/10.1109/lcomm.2019.2956500.

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Wang, Cong. "A Novel Model for Large-Scale Online College Learning in Postpandemic Era: AI-Driven Approach." Mobile Information Systems 2021 (December 14, 2021): 1–10. http://dx.doi.org/10.1155/2021/1048186.

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COVID-19 is a pandemic with a wide reach and explosive magnitude, and the world has been bracing itself for impact. Many have lost their jobs and savings, and many are homeless. For better or worse, COVID-19 has permanently changed our lives. For college students, the pandemic means giving up most of the on-campus experience in the postpandemic era and performing online learning instead. Virtual lessons may become a permanent part of college education. Large-scale online learning typically utilizes interactive live video streaming. In this study, we analyzed a codec and video streaming transmission protocol using artificial intelligence. First, we studied an intraframe prediction optimization algorithm for the H.266 codec based on long short-term memory networks. In terms of video streaming transmission protocols, real-time communication optimization based on Quick UDP Internet connections and Luby Transform codes is proposed to improve the quality of interactive live video streaming. Experimental results demonstrate that the proposed strategy outperforms three benchmarks in terms of video streaming quality, video streaming latency, and average throughput.
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Badr, Ahmed, Pratik Patil, Ashish Khisti, Wai-Tian Tan, and John Apostolopoulos. "Layered Constructions for Low-Delay Streaming Codes." IEEE Transactions on Information Theory 63, no. 1 (January 2017): 111–41. http://dx.doi.org/10.1109/tit.2016.2618924.

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Yıldız, Ezgi Pelin, and Sahap Altınbas. "Investigation of Efficient Backup Tecniques To Reduce Late In Cloud Systems: A Modeling Study." Global Journal of Information Technology: Emerging Technologies 13, no. 1 (April 1, 2023): 46–54. http://dx.doi.org/10.18844/gjit.v13i1.8863.

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From the formation stages of the information society to the present day, new developments have been experienced in many areas such as data formation, processing, storage and sharing of data in electronic environments. In this context, many new technological concepts have been included in our lives. When it comes to data storage environments, the concept of "Cloud Computing", which has been developing and increasing in popularity in recent years, comes to our minds. Cloud computing, with its known definition, is an internet-based technology service that provides access to the data stored on its servers at any time through internet access and allows users to benefit from the services offered by the system to the extent they want. While the advantages of cloud computing systems such as low cost, high performance, not requiring physical materials and being flexible can be mentioned, it is also emphasized that it brings many problems in terms of system. The most important of these problems; the lack of resources in cloud computing service providers, causing unpredictable delays in server response times. Based on all this information, this research is structured on the development of solution processes on these delays. In this sense, it is possible to consider three areas of cloud infrastructure; cloud computing, distributed storage and streaming communications. One of the first solutions that comes to mind is duplicating a task on multiple machines and waiting for the earliest copy to finish can reduce service latency; but intuitively, it costs additional computing resources and increases the queue load on servers. In this context, the effect of redundancy on the tail will be analyzed in the first part of the research. As a solution proposal, in this section, it will be discussed that service delays and queue load can be reduced by using the replication method, thus making the system more efficient. In a similar way, by requesting more than one copy of a file and waiting for any of them to arrive, the cloud storage requests will be accelerated for content download. In the second part of the research, generalization will be made from replication to coding and the (n, k) fork-join model will be studied to analyze the delay in accessing a storage system with (n, k) deletion code. This analysis will provide practical information that several users can access a content at the same time and provide faster service to the relevant users. Achieving low latency in streaming communication is even more difficult, as packets must be transmitted quickly and sequentially. Based on this structure, in the third part of the research, it is aimed to develop deletion codes to transmit redundant packet combinations and ensure smooth playback. Experimental modeling method was used as a research method in the study. In general, the aim of the research is to blend various mathematical tools from queuing, theory coding and renewal processes. It is foreseen that the techniques and insights to be developed with these dimensions of the research can be applied to other systems with stochastically changing components, and integrated studies can be made and contribute to the literature in this context.
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Petracca, Matteo, Claudio Salvadori, Stefano Bocchino, and Paolo Pagano. "Error Resilient Video Streaming with BCH Code Protection in Wireless Sensor Networks." Journal of Communications Software and Systems 10, no. 1 (March 21, 2014): 41. http://dx.doi.org/10.24138/jcomss.v10i1.139.

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Video streaming in Wireless Sensor Networks (WSNs) is a promising and challenging application for enabling high-value services. In such a context, the reduced amount ofavailable bandwidth, as well as the low-computational power available for acquiring and processing video frames, imposes the transmission of low resolution images at a low frame rate. Considering the aforementioned limitations, the amount of information carried by each video frame must be considered of utmost importance and preserved, as much as possible, against network losses that could introduce possible artifacts in the reconstructed dynamics of the scene.In this paper we first evaluate the impact of the bit error rate on the quality of the received video stream in a real scenario, then we propose a forward error correction technique based on the use of BCH codes with the aim of preserving the video quality. The proposed technique, against already proposed techniques in the WSN research field, has been specially designed to maintain a full back-compatibility with the IEEE802.15.4 standard in order to create a suitable solution aiming at accomplishing the Internet of Things (IoT) vision. Performance results evaluated in terms of Peak Signal-to-Noise Ratio (PSNR) show that the proposed solution reaches a PSNR improvement of 4.16 dB with respect to an unprotected transmission, while requiring an additional overhead equal to 22.51% in number of transmitted bits, and minimal impact on frame rate reduction and energy consumption. When higher protection levels have been imposed, bigger PSNR values have been experienced at the cost of an increased additional overhead, lower frame rates, and bigger energy consumption values.
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Okwudire, Chinedum, Sharankumar Huggi, Sagar Supe, Chengyang Huang, and Bowen Zeng. "Low-Level Control of 3D Printers from the Cloud: A Step toward 3D Printer Control as a Service." Inventions 3, no. 3 (August 19, 2018): 56. http://dx.doi.org/10.3390/inventions3030056.

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Control as a Service (CaaS) is an emerging paradigm where low-level control of a device is moved from a local controller to the Cloud, and provided to the device as an on-demand service. Among its many benefits, CaaS gives the device access to advanced control algorithms which may not be executable on a local controller due to computational limitations. As a step toward 3D printer CaaS, this paper demonstrates the control of a 3D printer by streaming low-level stepper motor commands (as opposed to high-level G-codes) directly from the Cloud to the printer. The printer is located at the University of Michigan, Ann Arbor, while its stepper motor commands are calculated using an advanced motion control algorithm running on Google Cloud computers in South Carolina and Australia. The stepper motor commands are sent over the internet using the user datagram protocol (UDP) and buffered to mitigate transmission delays; checks are included to ensure accuracy and completeness of the transmitted data. All but one part printed using the cloud-based controller in both locations were hitch free (i.e., no pauses due to excessive transmission delays). Moreover, using the cloud-based controller, the parts printed up to 54% faster than using a standard local controller, without loss of accuracy.
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Giraldo Barrada, Jorge Enrique, Juan Camilo García Viana, John Edison Morales Galeano, and Emanuel Valencia Henáo. "Construction of metal transfer modes maps for an ER4130 filler metal in GMAW process." DYNA 87, no. 215 (November 5, 2020): 126–35. http://dx.doi.org/10.15446/dyna.v87n215.86825.

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Metal transfer modes (MTMs) maps were constructed for GMAW process using ER4130 and 98%Ar-2%O2 shielding gas. There is no available MTMs maps for this filler metal which is used to obtain matching strength in welds of AISI 4130/4140 steels. These maps serve as tools to establish the MTM given a welding current and voltage, which is useful when an engineer is trying to qualify welding procedures according to construction codes. The maps were built analyzing current and voltage signals recorded at 5000 samples/second during bead-on-plate welds. The main advantage of this methodology is its simplicity of instrumentation without expensive cameras, but has low resolution and it is difficult to identify finer characteristics of MTMs, such as subgroups (repelled globular, streaming, rotational spray), drop diameter, explosive transfer, etc. Several MTMs were identified in the signal analysis and grouped into natural MTMs (short circuit, globular and spray) and interchangeable modes (short-circuit-globular, globular-spray and short-circuit-globular-spray).
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Shoucri, M., X. Lavocat-Dubuis, J. P. Matte, and F. Vidal. "Numerical study of ion acceleration and plasma jet formation in the interaction of an intense laser beam normally incident on an overdense plasma." Laser and Particle Beams 29, no. 3 (July 11, 2011): 315–32. http://dx.doi.org/10.1017/s026303461100036x.

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AbstractWe present a numerical study of the acceleration of ions in the interaction of a high intensity circularly polarized laser beam normally incident on an overdense plasma target, and the subsequent formation of neutral plasma ejected toward the rear side of the target. We compare the results obtained from two different numerical codes. We use an Eulerian Vlasov code for the numerical solution of the one-dimensional relativistic Vlasov-Maxwell set of equations, for both electrons and ions, and a particle-in-cell code applied to the same problem. We consider the case when the laser free space wavelength λ0 is greater than the scale length of the jump in the plasma density at the target plasma edge Ledge (λ0 ≫ Ledge), and the ratio of the plasma density to the critical density is such that n/ncr ≫ 1. The ponderomotive pressure due to the incident high-intensity laser radiation pushes the electrons at the target plasma surface, producing a sharp density gradient at the plasma surface, which gives rise to a charge separation. The resulting electric field accelerates the ions that reach a free streaming expansion phase, where they are neutralized by the electrons. A neutral plasma jet is thus ejected toward the rear side of the target. Two cases are studied: In the first case, the laser intensity rises to a maximum and then remains constant, and in the second case, the laser intensity is a Gaussian-shaped pulse. The results show substantial differences in the phase-space structure of the ions and the electrons between these two cases. There is good agreement between the quantitative macroscopic results obtained by the two codes, and good qualitative agreement between the results showing the kinetic details of the phase-space structures. The low noise level of the Eulerian Vlasov code allows a more detailed representation of the phase-space structures associated with this system, especially in the low density regions of the phase-space where ions are accelerated.
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Goel, Ashvin, Charles Krasic, and Jonathan Walpole. "Low-latency adaptive streaming over tcp." ACM Transactions on Multimedia Computing, Communications, and Applications 4, no. 3 (August 2008): 1–20. http://dx.doi.org/10.1145/1386109.1386113.

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10

Maiya, S. V., Daniel J. Costello, and T. E. Fuja. "Low Latency Coding: Convolutional Codes vs. LDPC Codes." IEEE Transactions on Communications 60, no. 5 (May 2012): 1215–25. http://dx.doi.org/10.1109/tcomm.2012.042712.110189.

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Evnitskaya, Natalia, and Ana Llinares. "(In)equity in CLIL programs?" AILA Review 35, no. 2 (December 31, 2022): 227–49. http://dx.doi.org/10.1075/aila.22026.evn.

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Abstract This study explores issues of potential (in)equity in Content and Language Integrated Learning (CLIL) programs in bilingual secondary schools in the Madrid region (Spain). Based on their general L2 proficiency, students in grades 7 to 10 are streamed into either High Exposure (HE) or Low Exposure (LE) strands, with different degrees of exposure to CLIL. Although this system ensures that all students in a bilingual secondary school receive CLIL to a certain degree, recent voices have signaled the potential risk of fostering inequality among students by streaming within the program (Fernández-Agüero & Hidalgo-McCabe, 2020; Hidalgo-McCabe, 2020). In this study, we explore classroom interactional practices by one science teacher teaching the same content in both groups (grade 7 HE and LE strands), and the effect of such interactional practices on enhancing (or not) students’ higher order thinking skills and the expression of academic content in the L2 or L1. For the analysis, we developed a multi-layered analytical model which incorporates the construct of cognitive discourse functions (CDFs) (Dalton-Puffer, 2013) and the semantic dimension of Legitimation Code Theory (LCT) (Maton, 2013, 2020). We find significant differences across the two groups in the use of CDFs and ‘semantic codes’ for knowledge construction and meaning making. More specifically, the results show a more frequent use of the CDF evaluate and a higher rate of semantic density (abstractions) in classroom discourse in the HE strand.
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Kinoshita, Toya, and Hiroyuki Hisamatsu. "Low Latency Live Streaming System with Congestion Control." Journal of Advances in Computer Networks 9, no. 1 (2021): 8–13. http://dx.doi.org/10.18178/jacn.2021.9.1.280.

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In recent years, the traffic for live streaming on the web has been increasing. The current live streaming methods that use MPEG-DASH or HLS are simple and scale easily to many clients using HTTP. However, they do not take into account the communication between the distributor and the viewer. As a result, latency between the distributor and the viewer is relatively high. Therefore, in this paper, we propose a low latency live streaming system on the web using WebRTC. Since WebRTC uses UDP, it does not have a congestion control mechanism. Depending on the network congestion, it is possible to stream video with quality that exceeds the available bandwidth. Therefore, we propose a system to change the video quality based on the congestion status. The proposed system increases or decreases the video transfer rate by changing the quality of the streamed video depending on the network conditions. We have evaluated the proposed system in a real network environment. As a result, we showed that the delay of the proposed system is smaller than that of the MPEG-DASH system. We also showed that the proposed system can change the quality of the video and switch the transmission rate appropriately according to the network conditions.
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Jiang, Yihan, Hyeji Kim, Himanshu Asnani, Sreeram Kannan, Sewoong Oh, and Pramod Viswanath. "LEARN Codes: Inventing Low-Latency Codes via Recurrent Neural Networks." IEEE Journal on Selected Areas in Information Theory 1, no. 1 (May 2020): 207–16. http://dx.doi.org/10.1109/jsait.2020.2988577.

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14

Zhang, Juntan, Jonathan S. Yedidia, and Marc P. C. Fossorier. "Low-Latency Decoding of EG LDPC Codes." Journal of Lightwave Technology 25, no. 9 (September 2007): 2879–86. http://dx.doi.org/10.1109/jlt.2007.903311.

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15

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.
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Bichot, Guillaume, and Nicolas Le Scouarnec. "Server-Side Segment Selection for Low-Latency Streaming—S4S." SMPTE Motion Imaging Journal 130, no. 10 (November 2021): 50–56. http://dx.doi.org/10.5594/jmi.2021.3117125.

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Miao, Haoran, Gaofeng Cheng, and Pengyuan Zhang. "Low‐latency transformer model for streaming automatic speech recognition." Electronics Letters 58, no. 1 (October 27, 2021): 44–46. http://dx.doi.org/10.1049/ell2.12349.

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Wu, Eric Hsiao-Kuang, Ming-I. Hsieh, and Hsu-Te Lai. "Low latency and efficient packet scheduling for streaming applications." Computer Communications 29, no. 9 (May 2006): 1413–21. http://dx.doi.org/10.1016/j.comcom.2005.09.001.

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Celebi, Kerem C., Shannon K. T. Bailey, Micheal W. Burns, and Kunal Bansal. "Is Virtual Reality Streaming Ready for Remote Medical Education? Measuring Latency of Stereoscopic VR for Telementoring." Proceedings of the Human Factors and Ergonomics Society Annual Meeting 65, no. 1 (September 2021): 757–61. http://dx.doi.org/10.1177/1071181321651332.

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Telementoring in healthcare education has been used successfully to teach technical skills and clinical reasoning when in-person instruction is not feasible; however, previous technology for telementoring had limitations such as narrow field-of-view and high latency. Novel virtual reality (VR) livestreaming technology may address issues in traditional 2-dimensional (2D) systems by expanding the field of view while streaming with low latency. Low latency streaming of video and audio is necessary for smooth communication between a medical specialist and remote trainees. If latency is low between the instructor and the remote trainees, conversations can be held without a noticeable delay, supporting synchronous instruction and collaboration. This research reports the first latency test results of a novel VR system that livestreams stereoscopic video and audio to remote VR headsets. Results showed the one-way audio and video latency was less than half a second, confirming the viability of live VR for medical telementoring.
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Das, Abhishek, and Nur A. Touba. "A Single Error Correcting Code with One-Step Group Partitioned Decoding Based on Shared Majority-Vote." Electronics 9, no. 5 (April 26, 2020): 709. http://dx.doi.org/10.3390/electronics9050709.

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Technology scaling has led to an increase in density and capacity of on-chip caches. This has enabled higher throughput by enabling more low latency memory transfers. With the reduction in size of SRAMs and development of emerging technologies, e.g., STT-MRAM, for on-chip cache memories, reliability of such memories becomes a major concern. Traditional error correcting codes, e.g., Hamming codes and orthogonal Latin square codes, either suffer from high decoding latency, which leads to lower overall throughput, or high memory overhead. In this paper, a new single error correcting code based on a shared majority voting logic is presented. The proposed codes trade off decoding latency in order to improve the memory overhead posed by orthogonal Latin square codes. A latency optimization technique is also proposed which lowers the decoding latency by incurring a slight memory overhead. It is shown that the proposed codes achieve better redundancy compared to orthogonal Latin square codes. The proposed codes are also shown to achieve lower decoding latency compared to Hamming codes. Thus, the proposed codes achieve a balanced trade-off between memory overhead and decoding latency, which makes them highly suitable for on-chip cache memories which have stringent throughput and memory overhead constraints.
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van der Hooft, Jeroen, Cedric De Boom, Stefano Petrangeli, Tim Wauters, and Filip De Turck. "Performance Characterization of Low-Latency Adaptive Streaming From Video Portals." IEEE Access 6 (2018): 43039–55. http://dx.doi.org/10.1109/access.2018.2863033.

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Liu, Leslie S., and Roger Zimmermann. "Adaptive low-latency peer-to-peer streaming and its application." Multimedia Systems 11, no. 6 (April 12, 2006): 497–512. http://dx.doi.org/10.1007/s00530-006-0030-4.

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Minja, Aleksandar, Dušan Dobromirov, and Vojin Šenk. "Bidirectional stack decoding of polar codes." Vojnotehnicki glasnik 69, no. 2 (2021): 405–15. http://dx.doi.org/10.5937/vojtehg69-29858.

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Introduction/purpose: The paper introduces a reduced latency stack decoding algorithm of polar codes, inspired by the bidirectional stack decoding of convolutional codes and based on the folding technique. Methods: The stack decoding algorithm (also known as stack search) that is useful for decoding tree codes, the list decoding technique introduced by Peter Elias and the folding technique for polar codes which is used to reduce the latency of the decoding algorithm. The simulation was done using the Monte Carlo procedure. Results: A new polar code decoding algorithm, suitable for parallel implementation, is developed and the simulation results are presented. Conclusions: Polar codes are a class of capacity achieving codes that have been adopted as the main coding scheme for control channels in 5G New Radio. The main decoding algorithm for polar codes is the successive cancellation decoder. This algorithm performs well at large blocklengths with a low complexity, but has very low reliability at short and medium blocklengths. Several decoding algorithms have been proposed in order to improve the error correcting performance of polar codes. The successive cancellation list decoder, in conjunction with a cyclic redundancy check, provides very good error-correction performance, but at the cost of a high implementation complexity. The successive cancellation stack decoder provides similar error-correction performance at a lower complexity. Future machine-type and ultra reliable low latency communication applications require high-speed low latency decoding algorithms with good error correcting performance. In this paper, we propose a novel decoding algorithm, inspired by the bidirectional stack decoding of classical convolutional codes, with reduced latency that achieves similar performance as the classical successive cancellation list and successive cancellation stack decoding algorithms. The results are presented analytically and verified by simulation.
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Li, Wenjie, Jing Tian, Jun Lin, and Zhongfeng Wang. "Modified GII-BCH Codes for Low-Complexity and Low-Latency Encoders." IEEE Communications Letters 23, no. 5 (May 2019): 785–88. http://dx.doi.org/10.1109/lcomm.2019.2908867.

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Jali, N., P. Muralidhar, and S. R. Patri. "Low Latency SC Decoder Architecture for Interleaved Polar Codes." Radioengineering 31, no. 3 (September 2022): 398–405. http://dx.doi.org/10.13164/re.2022.0398.

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Liu, Jingjing, and Rodrigo C. de Lamare. "Low-Latency Reweighted Belief Propagation Decoding for LDPC Codes." IEEE Communications Letters 16, no. 10 (October 2012): 1660–63. http://dx.doi.org/10.1109/lcomm.2012.080312.121307.

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Qin, Kangjian, and Zhaoyang Zhang. "Low-Latency Adaptive Ordered Statistic Decoding of Polar Codes." IEEE Access 7 (2019): 134226–35. http://dx.doi.org/10.1109/access.2019.2940525.

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Gazi, Orhan, and Ahmed A. S. Al-Kattan. "Low Latency Hybrid Structures Involving Convolutional Product like Codes and Space Time Codes." Wireless Personal Communications 94, no. 3 (August 30, 2016): 745–54. http://dx.doi.org/10.1007/s11277-016-3648-3.

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El-Abbasy, Karim, Ramy Taki Eldin, Salwa El Ramly, and Bassant Abdelhamid. "Optimized Polar Codes as Forward Error Correction Coding for Digital Video Broadcasting Systems." Electronics 10, no. 17 (September 3, 2021): 2152. http://dx.doi.org/10.3390/electronics10172152.

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Polar codes are featured by their low encoding/decoding complexity for symmetric binary input-discrete memoryless channels. Recently, flexible generic Successive Cancellation List (SCL) decoders for polar codes were proposed to provide different throughput, latency, and decoding performances. In this paper, we propose to use polar codes with flexible fast-adaptive SCL decoders in Digital Video Broadcasting (DVB) systems to meet the growing demand for more bitrates. In addition, they can provide more interactive services with less latency and more throughput. First, we start with the construction of polar codes and propose a new mathematical relation to get the optimized design point for the polar code. We prove that our optimized design point is too close to the one that achieves minimum Bit Error Rate (BER). Then, we compare the performance of polar and Low-Density Parity Check (LDPC) codes in terms of BER, encoder/decoder latencies, and throughput. The results show that both channel coding techniques have comparable BER. However, polar codes are superior to LDPC in terms of decoding latency, and system throughput. Finally, we present the possible performance enhancement of DVB systems in terms of decoding latency and complexity when using optimized polar codes as a Forward Error Correction (FEC) technique instead of Bose Chaudhuri Hocquenghem (BCH) and LDPC codes that are currently adopted in DVB standards.
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Zhang, Bo, Thiago Teixeira, and Yuriy Reznik. "Performance of Low-Latency DASH and HLS Streaming in Mobile Networks." SMPTE Motion Imaging Journal 131, no. 7 (August 2022): 26–34. http://dx.doi.org/10.5594/jmi.2022.3180777.

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Xiao, Yilin, Liang Xiao, Zefang Lv, Guohang Niu, Yuzhen Ding, and Wenyuan Xu. "Learning-Based Low-Latency VIoT Video Streaming Against Jamming and Interference." IEEE Wireless Communications 28, no. 4 (August 2021): 12–18. http://dx.doi.org/10.1109/mwc.101.2000474.

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Yahia, Mariem Ben, Yannick Le Louedec, Gwendal Simon, Loutfi Nuaymi, and Xavier Corbillon. "HTTP/2-based Frame Discarding for Low-Latency Adaptive Video Streaming." ACM Transactions on Multimedia Computing, Communications, and Applications 15, no. 1 (February 25, 2019): 1–23. http://dx.doi.org/10.1145/3280854.

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Marai, Oussama El, and Tarik Taleb. "Smooth and Low Latency Video Streaming for Autonomous Cars During Handover." IEEE Network 34, no. 6 (November 2020): 302–9. http://dx.doi.org/10.1109/mnet.011.2000258.

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Lobov, Igor V., and Vladislav G. Gotman. "LOW LATENCY SEAMLESS LIVE TCP-STREAMING BY MEANS OF PROGRESSIVE DOWNLOAD METHOD." T-Comm 15, no. 1 (2021): 11–18. http://dx.doi.org/10.36724/2072-8735-2021-15-1-11-18.

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The article discusses the client-server technology for organizing seamless live TCP-streaming with the use of progressive download method. Main factors which have a major impact on the overall playback latency have been identified. A study on the most important factor – the size of the client’s receiving buffer Lbuf – was carried out. It was found that the hypothetical display time of the last frame taken consists of two components: 1) the general instability, which has the character of a pronounced “band of instability”, and 2) episodic “drawdowns” associated with spontaneous media flow retentions in data transmission channel.The result of these drawdowns is an increase in Lbuf, resulting in increased playback delay. The article proposes a method to reduce the client’s receive buffer latency downto acceptable level (100-300 ms), allowing two-way communication between clients – an algorithm for smooth correction of client receiving buffer size. The article includes the results of the smooth correction algorithm in the implementation of TCP-Streaming by means of progressive downloading method. The implementation is a distributed software package consisting of one or more encoding processes, retransmitting process, client-side software. The media-flow Encoders converts data from the media-data sources (camera, computer screen, microphone) and transmits them to the clients via Retransmitter. The client-side software is a set of web pages (HTML + JS) where the receive buffer smooth correction algorithm is implemented. It was concluded that live TCP-streaming by means of progressive download method is quite suited to organize the communication with low latency between clients.
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Siva Kumar, M., S. Syed Shameem, M. N. V. Raghu Sai, Dheeraj Nikhil, P. Kartheek, and K. Hari Kishore. "Efficient and low latency turbo encoder design using Verilog-Hdl." International Journal of Engineering & Technology 7, no. 1.5 (December 31, 2017): 37. http://dx.doi.org/10.14419/ijet.v7i1.5.9119.

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Low complexity turbo-like codes based totally on the simple trellis or simple graph shape consequences in encoding with low complexity. Out of this Convolution, encoder and turbo codes are widely used due to the splendid errors control performance. The most famous communications encoding set of rules, the iterative deciphering calls for an exponential expansion in hardware complexity to acquire expanded encode accuracy. This paper makes a usage of Log-Map based Iterative decoding technique and specialty in the conclusion of the turbo encoder. The rapid codes are designed with the help of Recursive Systematic Convolution and are separated thru interleave, which (thing used to rearrange the bit collection) plays an essential position within the encoding technique. This paper offers the design of the parallel connection of Recursive Systematic Convolution (RSC) encoders and interleave to restrict postpone, results to form a turbo Encoder. The turbo Encoder is designed by way of Verilog-HDL and Synthesized through Xilinx ISE
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36

Hu, Liang, Rui Sun, Feng Wang, Xiuhong Fei, and Kuo Zhao. "A Stream Processing System for Multisource Heterogeneous Sensor Data." Journal of Sensors 2016 (2016): 1–8. http://dx.doi.org/10.1155/2016/4287834.

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With the rapid development of the Internet of Things (IoT), a variety of sensor data are generated around everyone’s life. New research perspective regarding the streaming sensor data processing of the IoT has been raised as a hot research topic that is precisely the theme of this paper. Our study serves to provide guidance regarding the practical aspects of the IoT. Such guidance is rarely mentioned in the current research in which the focus has been more on theory and less on issues describing how to set up a practical system. In our study, we employ numerous open source projects to establish a distributed real time system to process streaming data of the IoT. Two urgent issues have been solved in our study that are (1) multisource heterogeneous sensor data integration and (2) processing streaming sensor data in real time manner with low latency. Furthermore, we set up a real time system to process streaming heterogeneous sensor data from multiple sources with low latency. Our tests are performed using field test data derived from environmental monitoring sensor data collected from indoor environment for system validation. The results show that our proposed system is valid and efficient for multisource heterogeneous sensor data integration and streaming data processing in real time manner.
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37

Sun, Zhen, Zhao Chen, Liuguo Yin, and Jianhua Lu. "Design of LDBCH Codes for Ultra Reliable Low Latency Communications." IEEE Communications Letters 25, no. 9 (September 2021): 2800–2804. http://dx.doi.org/10.1109/lcomm.2021.3092629.

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38

Shirvanimoghaddam, Mahyar, Mohammad Sadegh Mohammadi, Rana Abbas, Aleksandar Minja, Chentao Yue, Balazs Matuz, Guojun Han, et al. "Short Block-Length Codes for Ultra-Reliable Low Latency Communications." IEEE Communications Magazine 57, no. 2 (February 2019): 130–37. http://dx.doi.org/10.1109/mcom.2018.1800181.

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39

Hladky, Jozef, Michael Stengel, Nicholas Vining, Bernhard Kerbl, Hans-Peter Seidel, and Markus Steinberger. "QuadStream." ACM Transactions on Graphics 41, no. 6 (November 30, 2022): 1–13. http://dx.doi.org/10.1145/3550454.3555524.

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Streaming rendered 3D content over a network to a thin client device, such as a phone or a VR/AR headset, brings high-fidelity graphics to platforms where it would not normally possible due to thermal, power, or cost constraints. Streamed 3D content must be transmitted with a representation that is both robust to latency and potential network dropouts. Transmitting a video stream and reprojecting to correct for changing viewpoints fails in the presence of disocclusion events; streaming scene geometry and performing high-quality rendering on the client is not possible on limited-power mobile GPUs. To balance the competing goals of disocclusion robustness and minimal client workload, we introduce QuadStream , a new streaming content representation that reduces motion-to-photon latency by allowing clients to efficiently render novel views without artifacts caused by disocclusion events. Motivated by traditional macroblock approaches to video codec design, we decompose the scene seen from positions in a view cell into a series of quad proxies , or view-aligned quads from multiple views. By operating on a rasterized G-Buffer, our approach is independent of the representation used for the scene itself; the resulting QuadStream is an approximate geometric representation of the scene that can be reconstructed by a thin client to render both the current view and nearby adjacent views. Our technical contributions are an efficient parallel quad generation, merging, and packing strategy for proxy views covering potential client movement in a scene; a packing and encoding strategy that allows masked quads with depth information to be transmitted as a frame-coherent stream; and an efficient rendering approach for rendering our QuadStream representation into entirely novel views on thin clients. We show that our approach achieves superior quality compared both to video data streaming methods, and to geometry-based streaming.
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40

Mahdi Salih, Layla, Thuraya Mahmood Al-Qaradaghi, and Jalal Jamal Hamad Ameen. "An investigation of low-density parity-check codes and polar codes for future communication systems." Indonesian Journal of Electrical Engineering and Computer Science 30, no. 1 (April 1, 2023): 110. http://dx.doi.org/10.11591/ijeecs.v30.i1.pp110-120.

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In the fifth-generation (5G) era and future, mobile internet and internet of things (IoT) are the driving forces for mobile communications’ development. The three main 5G usage scenarios: enhanced mobile broadband (eMBB), massive machine-type communication (mMTC), and ultra-reliable and low-latency communications (URLLC), require improvement in throughput, reliability, and latency as compared with the previous fourth generation (4G) system. In this paper, an investigation is done on the coding part of the wireless communication systems. Two channel coding types; low-density parity-check (LDPC) code which is used as the coding scheme for data transmission, and Polar code which is utilized for control in 5G are discussed. Moreover, simulations are performed to assess their performance. The simulation results revealed the superiority of polar code for transmitting short information messages and LDPC for transmitting long data messages. The use of LDPC and polar codes in 5G communication systems is justified by their ability to accommodate a wide range of data lengths and code rates, as well as their good bit error rate (BER) performance. Furthermore, the effect of the number of iterations on the BER performance of LDPC code and different decoding algorithms of polar code are considered.
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41

Li, Ye, Feifan Zhang, Jue Wang, Tony Q. S. Quek, and Jiangzhou Wang. "On Streaming Coding for Low-Latency Packet Transmissions Over Highly Lossy Links." IEEE Communications Letters 24, no. 9 (September 2020): 1885–89. http://dx.doi.org/10.1109/lcomm.2020.2989367.

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42

Greco, Claudio, Marco Cagnazzo, and Béatrice Pesquet-Popescu. "Low-Latency Video Streaming With Congestion Control in Mobile Ad-Hoc Networks." IEEE Transactions on Multimedia 14, no. 4 (August 2012): 1337–50. http://dx.doi.org/10.1109/tmm.2012.2195480.

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43

Wang, Qianfan, Suihua Cai, Li Chen, and Xiao Ma. "Semi-LDPC Convolutional Codes: Construction and Low-Latency Windowed List Decoding." Journal of Communications and Information Networks 6, no. 4 (December 2021): 411–19. http://dx.doi.org/10.23919/jcin.2021.9663105.

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44

Jiang, Ming, and Dongli Fan. "A Low-Latency BF Decoding of LDPC Codes With Dynamic Thresholds." IEEE Communications Letters 25, no. 9 (September 2021): 2781–85. http://dx.doi.org/10.1109/lcomm.2021.3088579.

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45

Feng, Baoping, and Rongke Liu. "Efficient-Memory and Low-Latency BP Decoding Algorithm for Polar Codes." IEEE Communications Letters 24, no. 6 (June 2020): 1236–39. http://dx.doi.org/10.1109/lcomm.2020.2982643.

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46

Argon, C., and S. W. McLaughlin. "A parallel decoder for low latency decoding of turbo product codes." IEEE Communications Letters 6, no. 2 (February 2002): 70–72. http://dx.doi.org/10.1109/4234.984698.

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47

Fan, YouZhe, ChenYang Xia, Ji Chen, Chi-Ying Tsui, Jie Jin, Hui Shen, and Bin Li. "A Low-Latency List Successive-Cancellation Decoding Implementation for Polar Codes." IEEE Journal on Selected Areas in Communications 34, no. 2 (February 2016): 303–17. http://dx.doi.org/10.1109/jsac.2015.2504318.

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48

Papadias, Serafeim, Zoi Kaoudi, Jorge-Arnulfo Quiané-Ruiz, and Volker Markl. "Space-efficient random walks on streaming graphs." Proceedings of the VLDB Endowment 16, no. 2 (October 2022): 356–68. http://dx.doi.org/10.14778/3565816.3565835.

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Graphs in many applications, such as social networks and IoT, are inherently streaming, involving continuous additions and deletions of vertices and edges at high rates. Constructing random walks in a graph, i.e., sequences of vertices selected with a specific probability distribution, is a prominent task in many of these graph applications as well as machine learning (ML) on graph-structured data. In a streaming scenario, random walks need to constantly keep up with the graph updates to avoid stale walks and thus, performance degradation in the downstream tasks. We present Wharf, a system that efficiently stores and updates random walks on streaming graphs. It avoids a potential size explosion by maintaining a compressed, high-throughput, and low-latency data structure. It achieves (i) the succinct representation by coupling compressed purely functional binary trees and pairing functions for storing the walks, and (ii) efficient walk updates by effectively pruning the walk search space. We evaluate Wharf, with real and synthetic graphs, in terms of throughput and latency when updating random walks. The results show the high superiority of Wharf over inverted index- and tree-based baselines.
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49

Sidorenko, V., W. F. J. Müller, W. Zabolotny, I. Fröhlich, D. Emschermann, and J. Becker. "Evaluation of GBT-FPGA for timing and fast control in CBM experiment." Journal of Instrumentation 18, no. 02 (February 1, 2023): C02052. http://dx.doi.org/10.1088/1748-0221/18/02/c02052.

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Abstract Timing and Fast Control (TFC) system for the Compressed Baryonic Matter (CBM) experiment is being developed with focus on low and deterministic data transmission latency. This helps to minimize data corruption in the free-streaming Data Acquisition (DAQ) system during occasional data bursts caused by the expected beam intensity fluctuations. Proven in latency-optimized experimental data transport applications, the GBT-FPGA core is expected to positively contribute to the TFC system performance. In this work, the core has been integrated as the primary communication interface and its effect on transmission latency and quality of time distribution has been evaluated.
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50

Lee, Seunghwa, and Joon Yoo. "Reinforcement Learning Based Multipath QUIC Scheduler for Multimedia Streaming." Sensors 22, no. 17 (August 23, 2022): 6333. http://dx.doi.org/10.3390/s22176333.

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With the recent advances in computing devices such as smartphones and laptops, most devices are equipped with multiple network interfaces such as cellular, Wi-Fi, and Ethernet. Multipath TCP (MPTCP) has been the de facto standard for utilizing multipaths, and Multipath QUIC (MPQUIC), which is an extension of the Quick UDP Internet Connections (QUIC) protocol, has become a promising replacement due to its various advantages. The multipath scheduler, which determines the path to which each packet should be transmitted, is a key function that affects the multipath transport performance. For example, the default minRTT scheduler typically achieves good throughput, while the redundant scheduler gains low latency. While the legacy schedulers may generally give a desirable performance in some environments, however, each application renders different requirements. For example, Web applications target low latency, while video streaming applications require low jitter and high video quality. In this paper, we propose a novel MPQUIC scheduler based on deep reinforcement learning using the Deep Q-Network (DQN) that enhances the quality of multimedia streaming. Our proposal first takes into account both delay and throughput as a reward for reinforcement learning to achieve a low video chunk download time. Second, we propose a chunk manager that informs the scheduler of the video chunk information, and we also tune the learning parameters to explore new random actions adequately. Finally, we implement our new scheduler on the Linux kernel and give results using the Mininet experiments. The evaluation results show that our proposal outperforms legacy schedulers by at least 20%.
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