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Relevant bibliographies by topics / Codes for low-latency streaming

Academic literature on the topic 'Codes for low-latency streaming'

Author: Grafiati

Published: 6 September 2023

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

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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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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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Dissertations / Theses on the topic "Codes for low-latency streaming"

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Goel, Ashvin. "Operating system support for low-latency streaming /." Full text open access at:, 2003. http://content.ohsu.edu/u?/etd,194.

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Tay, Kah Keng. "Low-latency network coding for streaming video multicast." Thesis, Massachusetts Institute of Technology, 2008. http://hdl.handle.net/1721.1/46523.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2008.
Includes bibliographical references (p. 95-98).
Network coding has been successfully employed to increase throughput for data transfers. However, coding inherently introduces packet inter-dependencies and adds decoding delays which increase latency. This makes it difficult to apply network coding to real-time video streaming where packets have tight arrival deadlines. This thesis presents FLOSS, a wireless protocol for streaming video multicast. At the core of FLOSS is a novel network code. This code maximizes the decoding opportunities at every receiver, and at the same time minimizes redundancy and decoding latency. Instead of sending packets plainly to a single receiver, a sender mixes in packets that are immediately beneficial to other receivers. This simple technique not only allows us to achieve the coding benefits of increased throughput, it also decreases delivery latency, unlike other network coding approaches. FLOSS performs coding over a rolling window of packets from a video flow, and determines with feedback the optimal set of packet transmissions needed to get video across in a timely and reliable manner. A second important characteristic of FLOSS is its ability to perform both interand intra-flow network coding at the same time. Our technique extends easily to support multiple video streams, enabling us to effectively and transparently apply network coding and opportunistic routing to video multicast in a wireless mesh. We devise VSSIM*, an improved video quality metric based on [46]. Our metric addresses a significant limitation of prior art and allows us to evaluate video with streaming errors like skipped and repeated frames. We have implemented FLOSS using Click [22]. Through experiments on a 12-node testbed, we demonstrate that our protocol outperforms both a protocol that does not use network coding and one that does so naively. We show that the improvement in video quality comes from increased throughput, decreased latency and opportunistic receptions from our scheme.
by Kah Keng Tay.
M.Eng.

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Tafleen, Sana. "Fault Tolerance Strategies for Low-Latency Live Video Streaming." Thesis, University of Louisiana at Lafayette, 2019. http://pqdtopen.proquest.com/#viewpdf?dispub=13420002.

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This paper describes the effect of failures on various video QoS metrics like delay, packet loss, and recovery time. SDN network has been used to guarantee reliability and efficient data transmission. There are many failures that can occur within the SDN mesh network or between the non-SDN and the SDN network. There is a need for both reliable and low-latency transmission of live video streams, especially in situations such as public safety or public gathering events. This is because everyone is trying to use the limited network at the same time. That leads to oversubscription and network outages, and computing devices may fail. Existing mechanisms built into TCP/IP and video streaming protocols, and fault tolerance strategies (such as buffering), are inadequate due to low latency and reliability requirements for live streaming, especially in the presence of limited bandwidth and computational power of mobile or edge devices. The objective of this paper is to develop an efficient fault tolerant strategy at the source-side to produce a high-quality video with low latency and data loss. To recover the lost data during failures, buffering approach is used to store chunks in a buffer and retransmit the lost frames, requested by the receiver.

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Ben, Yahia Mariem. "Low latency video streaming solutions based on HTTP/2." Thesis, Ecole nationale supérieure Mines-Télécom Atlantique Bretagne Pays de la Loire, 2019. http://www.theses.fr/2019IMTA0136/document.

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Les techniques adaptatives de transmission vidéo s’appuient sur un contenu qui est encodé à différents niveaux de qualité et divisé en segments temporels. Avant de télécharger un segment, le client exécute un algorithme d’adaptation pour décider le meilleur niveau de qualité à considérer. Selon les services, ce niveau de qualité doit correspondre aux ressources réseaux disponibles, mais aussi à d’autres éléments comme le mouvement de tête d’un utilisateur regardant une vidéo immersive (à 360°) afin de maximiser la qualité de la portion de la vidéo qui est regardée. L’efficacité de l’algorithme d’adaptation a un impact direct sur la qualité de l’expérience finale. En cas de mauvaise sélection de segment, un client HTTP/1 doit attendre le téléchargement du prochain segment afin de choisir une qualité appropriée. Dans cette thèse, nous proposons d’utiliser le protocole HTTP/2 pour remédier à ce problème. Tout d’abord, nous nous focalisons sur le service de vidéo en direct. Nous concevons une stratégie de rejet d’images vidéo quand la bande passante est très variable afin d’éviter les arrêts fréquents de la lecture vidéo et l’accumulation des retards. Le client doit demander chaque image vidéo dans un flux HTTP/2 dédié pour contrôler la livraison des images par appel aux fonctionnalités HTTP/2 au niveau des flux concernées. Ensuite, nous optimisons la livraison des vidéos immersives en bénéficiant de l’amélioration de la prédiction des mouvements de têtes de l’utilisateur grâce aux fonctionnalités d’initialisation et de priorité de HTTP/2. Les résultats montrent que HTTP/2 permet d’optimiser l’utilisation des ressources réseaux et de s’adapter aux latences exigées par chaque service
Adaptive video streaming techniques enable the delivery of content that is encoded at various levels of quality and split into temporal segments. Before downloading a segment, the client runs an adaptation algorithm to determine the level of quality that best matches the network resources. For immersive video streaming this adaptation mechanism should also consider the head movement of a user watching the 360° video to maximize the quality of the viewed portion. However, this adaptation may suffer from errors, which impact the end user’s quality of experience. In this case, an HTTP/1 client must wait for the download of the next segment to choose a suitable quality. In this thesis, we propose to use the HTTP/2 protocol instead to address this problem. First, we focus live streaming video. We design a strategy to discard video frames when the band width is very variable in order so as to avoid the rebuffering events and the accumulation of delays. The customer requests each video frame in an HTTP/2 stream which allows to control the delivery of frames by leveraging the HTTP/2 features at the level of the dedicated stream. Besides, we use the priority and reset stream features of HTTP/2 to optimize the delivery of immersive videos. We propose a strategy to benefit from the improvement of the user’s head movements prediction overtime. The results show that HTTP/2 allows to optimize the use of network resources and to adapt to the latencies required by each service

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Tideström, Jakob. "Investigation into low latency live video streaming performance of WebRTC." Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-249446.

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As WebRTC is intended for peer-to-peer real time communications, it contains the capability for streaming video at low latencies. This thesis leverages this ability to stream live video footage in a client-server scenario. Using a local broadcaster, server, and client setup, a static video file is streamed as live footage. The performance is compared with contemporary live streaming techniques, HTTP Live Streaming and Dynamic Adaptive Streaming over HTTP, streaming the same content. It is determined that WebRTC achieves lower latencies than both techniques.
Eftersom WebRTC är menat för peer-to-peer realtidskommunikation så har den förmågan att strömma video med låg latens. Denna avhandling utnyttjar den här förmågan för att strömma livevideo i ett klient-server-scenario. Med en uppsättning som omfattar en lokal sändare, server, och klient strömmas en statisk videofil som en live-video. Prestandan jämförs med hur de samtida liveströmningsteknikerna HTTP Live Streaming respective Dynamic Adaptive Streaming over HTTP strömmar samma innehåll. Slutsatsen är att WebRTC lyckas uppnå lägre latens än båda de andra teknikerna men utan relativt mycket finjustering så försämras kvaliteten på strömmen.

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Bhat, Amit. "Low-latency Estimates for Window-Aggregate Queries over Data Streams." PDXScholar, 2011. https://pdxscholar.library.pdx.edu/open_access_etds/161.

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Obtaining low-latency results from window-aggregate queries can be critical to certain data-stream processing applications. Due to a DSMS's lack of control over incoming data (typically, because of delays and bursts in data arrival), timely results for a window-aggregate query over a data stream cannot be obtained with guarantees about the results' accuracy. In this thesis, I propose a technique, which I term prodding, to obtain early result estimates for window-aggregate queries over data streams. The early estimates are obtained in addition to the regular query results. The proposed technique aims to maximize the contribution to a result-estimate computation from all the stateful operators across a multi-level query plan. I evaluate the benefits of prodding using real-world and generated data streams having different patterns in data arrival and data values. I conclude that, in various DSMS applications, prodding can generate low-latency estimates to window-aggregate query results. The main factors affecting the degree of inaccuracy in such estimates are: the aggregate function used in a query, the patterns in arrivals and values of stream data, and the aggressiveness of demanding the estimates. The utility of the estimates obtained using prodding should be optimized by tuning the aggressiveness in result-estimate demands to the specific latency and accuracy needs of a business, considering any available knowledge about patterns in the incoming data.

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Gazi, Orhan. "Parallelized Architectures For Low Latency Turbo Structures." Phd thesis, METU, 2007. http://etd.lib.metu.edu.tr/upload/12608110/index.pdf.

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In this thesis, we present low latency general concatenated code structures suitable for parallel processing. We propose parallel decodable serially concatenated codes (PDSCCs) which is a general structure to construct many variants of serially concatenated codes. Using this most general structure we derive parallel decodable serially concatenated convolutional codes (PDSCCCs). Convolutional product codes which are instances of PDSCCCs are studied in detail. PDSCCCs have much less decoding latency and show almost the same performance compared to classical serially concatenated convolutional codes. Using the same idea, we propose parallel decodable turbo codes (PDTCs) which represent a general structure to construct parallel concatenated codes. PDTCs have much less latency compared to classical turbo codes and they both achieve similar performance. We extend the approach proposed for the construction of parallel decodable concatenated codes to trellis coded modulation, turbo channel equalization, and space time trellis codes and show that low latency systems can be constructed using the same idea. Parallel decoding operation introduces new problems in implementation. One such problem is memory collision which occurs when multiple decoder units attempt accessing the same memory device. We propose novel interleaver structures which prevent the memory collision problem while achieving performance close to other interleavers.

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Sonono, Tofik. "Interoperable Retransmission Protocols with Low Latency and Constrained Delay : A Performance Evaluation of RIST and SRT." Thesis, KTH, Kommunikationssystem, CoS, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-254897.

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The media industry has during the last decade migrated services from dedicated medianetworks to more shared resources and lately also the public internet and public data centers. Inorder to cater for such transition, several protocols have been designed to meet the demand forhigh-quality media transport over lossy infrastructure, protocols such as SRT and RIST. Thepurpose of Reliable Internet Stream Transport (RIST) and Secure Reliable Transport (SRT) is tohave all vendors of broadcasting equipment support an interoperable way of communication. Thelack of interoperability locks consumers into one particular vendor’s family of products - most oftenthis equipment only supports a proprietary technology. Interoperability creates a more competitivemarket space which benefits consumers and gives vendors an incentive to be more innovative intheir solutions. The purpose of this thesis is to assess the performance of these protocols by comparing theirperformance to a proprietary solution (named ÖÖÖ in this thesis and seen as an establishedsolution in the industry). The challenge is to test these protocols in a lab environment, but have theresults represent real-world use. For this, a large subset of samples is needed along with samplesmeasured over a long period. This sampling was made possible by writing a script which automatesthe sampling process. The results indicate that the versions of RIST and SRT tested in this thesis to some extentcompare well to the selected established protocol (ÖÖÖ). In many scenarios, SRT even did muchbetter, mainly when a line with a single feed was tested. For instance, when the network suffered a2% drop rate and utilized retransmission SRT performed the best and was the only protocol whichhad some samples where no packets were dropped during one hour of measurements. Whenrunning all three protocols at the same time, SRT also did the best in a network with up to 12% droprate. The results in this thesis should give a broadcaster an idea of which of these protocols willfulfill their requirements in a broadcast application.
I mediabranschen finns det en efterfrågan på utrustning som har inslag av interoperabilitet.Anledningen till detta är att någon som köper produkter från en viss återförsäljare inte vill låsas in idenna återförsäljares ”ekosystem” i flera år framöver. Då en studio sällan uppgraderar hela sinproduktionskedja på samma gång ger interoperabilitet möjligheten att köpa utrustning från andraåterförsäljare när man ska uppgradera något i produktionslinan. Detta leder till en merkonkurrenskraftig marknad samt ger incentiv till nya innovativa lösningar. Detta examensarbete går ut på att utvärdera lösningar som tagits fram för att främjainteroperabilitet och jämföra dem med en existerande proprietärlösning. Reliable Internet StreamTransport (RIST) och Secure Reliable Transport (SRT) är två protokoll som tagits fram för just dettasyfte. Utmaningen med att utvärdera dessa protokoll är att i en labbmiljö få resultat som reflekteraranvändandet av protokollen i verkligheten. Detta har gjorts med hjälp av ett program som tagitsfram i detta examensarbete. Med detta program har testandet kunnat automatiseras. Resultaten i detta examensarbete visar potential hos båda RIST och SRT. SRT är i vissascenarion till och med bättre än den proprietära lösningen. Protokollen visar något buggigtbeteende i vissa instanser, såsom att i vissa fal sluta fungera och inte kunna återgå till normalfunktion utan manuell interaktion. Allt som allt är dock protokollen i de flesta fallen testade i dettaexamensarbete ett godtyckligt alternativ till den jämförda proprietära lösningen.

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Lai, Hsu-Te, and 賴旭德. "Low Latency and Efficient Packet Scheduling for Streaming Applications." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/30454995901658632809.

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碩士
國立中央大學
資訊工程研究所
91
Adequate bandwidth allocations and strict delay requirements are critical for real time applications. Packet scheduling algorithms like Class Based Queue (CBQ), Nested Deficit Round Robin (Nested-DRR) are designed to ensure the bandwidth reservation function. However, they might cause unsteady packet latencies and introduce extra application handling overhead, such as allocating a large buffer for playing the media stream. High and unstable latency of packets might jeopardize the corresponding Quality of Service since real-time applications prefer low playback latency. Existing scheduling algorithms which keep latency of packets stable require knowing the details of individual flows. GPS (General Processor Sharing)-like algorithms does not consider the real behavior of a stream. A real stream is not perfectly smooth after forwarded by routers. GPS-like algorithms will introduce extra delay on the stream which is not perfectly smooth. This thesis presents an algorithm which provides low latency and efficient packet scheduling service for streaming applications called LLEPS.

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Huang, Ting-Chun, and 黃亭鈞. "Realizing Low Latency Real-Time Video Streaming Service with TCP." Thesis, 2015. http://ndltd.ncl.edu.tw/handle/34440272109505633081.

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碩士
國立臺灣海洋大學
資訊工程學系
103
Most real-time video streams are delivered using UDP. Compared against TCP, UDP does not have the head-of-line blocking effect, and therefore the performance does not drop dramatically due to packet losses. However, UDP does not offer a reliable packet delivery service, and it may not work in certain network setups including traffic shaping, firewall, and NAT. Researchers have attempted to solve the aforementioned problem using SCTP. However, the performance of SCTP on real-time video streaming is not clear, and it is not built-in for most off-the-shelf operating systems including both desktop and mobile OSes. As a result, it could not be a good choice for the demanding real-time multimedia streaming applications such as cloud gaming and video surveillance. Based on the observation, we proposed a real-time video streaming protocol design based on TCP, which is called multiple-flow TCP model. In this model, we leverage concurrent TCP flows to deliver multimedia streams. In addition to take the benefits of reliable packet delivery, the performance drop caused on packet losses can be mitigated and therefore improve the overall throughput. Our evaluation shows that the multiple-flow TCP model has a similar performance to UDP, and it offers the benefits of TCP and SCTP. We further conduct user studies to understand real user experiences regarding the performance of the proposed model. It also shows that the multiple-flow TCP model can perform better than TCP and SCTP in terms of real-timeliness and video quality.

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Book chapters on the topic "Codes for low-latency streaming"

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Giard, Pascal, Claude Thibeault, and Warren J. Gross. "Low-Latency Software Polar Decoders." In High-Speed Decoders for Polar Codes, 31–53. Cham: Springer International Publishing, 2017. http://dx.doi.org/10.1007/978-3-319-59782-9_3.

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Amini Salehi, Mohsen, and Xiangbo Li. "Low-Latency Delivery Networks for Multimedia Streaming." In Multimedia Cloud Computing Systems, 125–51. Cham: Springer International Publishing, 2021. http://dx.doi.org/10.1007/978-3-030-88451-2_7.

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Mouhoubi, Oualid, Charbel Abdel Nour, and Amer Baghdadi. "Low Latency Architecture Design for Decoding 5G NR Polar Codes." In Design and Architecture for Signal and Image Processing, 16–28. Cham: Springer International Publishing, 2022. http://dx.doi.org/10.1007/978-3-031-12748-9_2.

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Cappellari, Paolo, Mark Roantree, and Soon Ae Chun. "A Scalable Platform for Low-Latency Real-Time Analytics of Streaming Data." In Communications in Computer and Information Science, 1–24. Cham: Springer International Publishing, 2017. http://dx.doi.org/10.1007/978-3-319-62911-7_1.

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Carbone, Paris, and Vana Kalogeraki. "A Topologically-Aware Overlay Tree for Efficient and Low-Latency Media Streaming." In Lecture Notes of the Institute for Computer Sciences, Social Informatics and Telecommunications Engineering, 383–99. Berlin, Heidelberg: Springer Berlin Heidelberg, 2009. http://dx.doi.org/10.1007/978-3-642-10625-5_24.

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Chen, Si, Yuan Zhang, Huan Peng, and Jinyao Yan. "A Joint Bitrate and Buffer Control Scheme for Low-Latency Live Streaming." In Intelligence Science and Big Data Engineering. Big Data and Machine Learning, 369–80. Cham: Springer International Publishing, 2019. http://dx.doi.org/10.1007/978-3-030-36204-1_31.

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Drioli, Carlo, Claudio Allocchio, and Nicola Buso. "Networked Performances and Natural Interaction via LOLA: Low Latency High Quality A/V Streaming System." In Lecture Notes in Computer Science, 240–50. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-40050-6_21.

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Fleury, Martin, and Laith Al-Jobouri. "Data Partitioning." In Advances in Multimedia and Interactive Technologies, 118–58. IGI Global, 2016. http://dx.doi.org/10.4018/978-1-4666-8850-6.ch004.

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Abstract:

Data partitioning is a source-coding technique that has existed in one form or another in the standardized hybrid video codecs up to recent times. In essence, it is a method of prioritizing coding data, resulting in video layers that can be separately communicated across an error-prone network. The Chapter includes the background that has led to data partitioning being included in the standardized codecs. As this Chapter discusses, it differs from scalable video because the output from conventional, single-layer encoders can be converted to multi-layer form, rather than requiring specialist codec extensions. It is shown that the methods of forming the partitions so far employed are: dividing transformed, residual coefficients into two or more layers; and dividing coded data by function into headers, intra-, and inter-coded residuals to form three or more layers. It is also shown how layering naturally combines with protection by channel coding. Used as an error resilience tool, data partitioning presents a low overhead method, suitable for benign as well as bad channels. And in the three-layer variety, error concealment at the decoder can significantly aid the reconstruction of damaged video frames. The Chapter will be of particular interest to developers charged with making a mobile, low-latency, or interactive video streaming application robust, as they can select from the data-partitioning methods and apply them to open-source code of the recent High Efficiency Video Coding (HEVC) codec standard. Broadcast TV can also benefit from data partitioning. Developers of codecs additionally will find in this Chapter a guide to research and ideas about data partitioning which could be incorporated into future codecs.

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Raheel, Muhammad Salman, and Raad Raad. "Streaming Coded Video in P2P Networks." In Advances in Wireless Technologies and Telecommunication, 188–222. IGI Global, 2017. http://dx.doi.org/10.4018/978-1-5225-2113-6.ch009.

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This chapter discusses the state of the art in dealing with the resource optimization problem for smooth delivery of video across a peer to peer (P2P) network. It further discusses the properties of using different video coding techniques such as Scalable Video Coding (SVC) and Multiple Descriptive Coding (MDC) to overcome the playback latency in multimedia streaming and maintains an adequate quality of service (QoS) among the users. The problem can be summarized as follows; Given that a video is requested by a peer in the network, what properties of SVC and MDC can be exploited to deliver the video with the highest quality, least upload bandwidth and least delay from all participating peers. However, the solution to these problems is known to be NP hard. Hence, this chapter presents the state of the art in approximation algorithms or techniques that have been proposed to overcome these issues.

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Raheel, Muhammad Salman, and Raad Raad. "Streaming Coded Video in P2P Networks." In Research Anthology on Recent Trends, Tools, and Implications of Computer Programming, 1304–39. IGI Global, 2021. http://dx.doi.org/10.4018/978-1-7998-3016-0.ch060.

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Abstract:

This chapter discusses the state of the art in dealing with the resource optimization problem for smooth delivery of video across a peer to peer (P2P) network. It further discusses the properties of using different video coding techniques such as Scalable Video Coding (SVC) and Multiple Descriptive Coding (MDC) to overcome the playback latency in multimedia streaming and maintains an adequate quality of service (QoS) among the users. The problem can be summarized as follows; Given that a video is requested by a peer in the network, what properties of SVC and MDC can be exploited to deliver the video with the highest quality, least upload bandwidth and least delay from all participating peers. However, the solution to these problems is known to be NP hard. Hence, this chapter presents the state of the art in approximation algorithms or techniques that have been proposed to overcome these issues.

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Conference papers on the topic "Codes for low-latency streaming"

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Tonomura, Yoshihide, Daisuke Shirai, Masahiko Kitamura, Takayuki Nakachi, Tatsuya Fujii, and Hitoshi Kiya. "Low-Density Generator Matrix Codes for IP Packet Video Streaming with Backward Compatibility." In ICC 2011 - 2011 IEEE International Conference on Communications. IEEE, 2011. http://dx.doi.org/10.1109/icc.2011.5963201.

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Kalan, Reza Shokri, Reza Farahani, Emre Karsli, Christian Timmerer, and Hermann Hellwagner. "Towards low latency live streaming." In MMSys '22: 13th ACM Multimedia Systems Conference. New York, NY, USA: ACM, 2022. http://dx.doi.org/10.1145/3524273.3532904.

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Shuai, Yongtao, Manuel Gorius, and Thorsten Herfet. "Low-latency dynamic adaptive video streaming." In 2014 IEEE International Symposium on Broadband Multimedia Systems and Broadcasting (BMSB). IEEE, 2014. http://dx.doi.org/10.1109/bmsb.2014.6873486.

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Kim, Seohyang, Seungyoung Shin, and Joonseok Moon. "UDP-based Extremely Low Latency Streaming." In 2022 IEEE 19th Annual Consumer Communications & Networking Conference (CCNC). IEEE, 2022. http://dx.doi.org/10.1109/ccnc49033.2022.9700635.

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Zhu, Xiaoqing, Jiang Zhu, Rong Pan, Mythili Suryanarayana Prabhu, and Flavio Bonomi. "Cloud-assisted streaming for low-latency applications." In 2012 International Conference on Computing, Networking and Communications (ICNC). IEEE, 2012. http://dx.doi.org/10.1109/iccnc.2012.6167565.

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Karagkioules, Theo, Rufael Mekuria, Dirk Griffioen, and Arjen Wagenaar. "Online learning for low-latency adaptive streaming." In MMSys '20: 11th ACM Multimedia Systems Conference. New York, NY, USA: ACM, 2020. http://dx.doi.org/10.1145/3339825.3397042.

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Sun, Liyang, Tongyu Zong, Siquan Wang, Yong Liu, and Yao Wang. "Tightrope walking in low-latency live streaming." In MMSys '21: 12th ACM Multimedia Systems Conference. New York, NY, USA: ACM, 2021. http://dx.doi.org/10.1145/3458305.3463382.

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Sanchez, Yago, Edward Grinshpun, David Faucher, Thomas Schierl, and Sameer Sharma. "Low latency DASH based streaming over LTE." In 2014 IEEE Visual Communications and Image Processing Conference (VCIP). IEEE, 2014. http://dx.doi.org/10.1109/vcip.2014.7051489.

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Bentaleb, Abdelhak, Christian Timmerer, Ali C. Begen, and Roger Zimmermann. "Bandwidth prediction in low-latency chunked streaming." In the 29th ACM Workshop. New York, New York, USA: ACM Press, 2019. http://dx.doi.org/10.1145/3304112.3325611.

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Matri, Pierre, and Robert Ross. "Neon: Low-Latency Streaming Pipelines for HPC." In 2021 IEEE 14th International Conference on Cloud Computing (CLOUD). IEEE, 2021. http://dx.doi.org/10.1109/cloud53861.2021.00089.

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