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Abstrakt

The use of video streaming is constantly increasing. High-resolution video requires resources on both the sender and the receiver side. Many compression techniques can be utilized to compress the video and simultaneously maintain quality. The main goal of this paper is to provide an overview of video streaming and QoE. This paper describes the basic concepts and discusses existing methodologies to measure QoE. Subjective, objective, and video compression technologies are discussed. This review paper gathers the codec implementation developed by MPEG, Google, and Apple. This paper outlines the challenges and future research directions that should be considered in the measurement and assessment of the quality of experience for video services.
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Autorzy i Afiliacje

Syed Uddin
1
Mikolaj Leszczuk
1
Michal Grega
1

  1. AGH University of Krakow, Poland
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Abstrakt

The paper presents the results of research and analysis of voice data transmission quality in IP packet networks. It analyses mechanisms allowing for the assessment of packet telephony data transmission quality. Possible transmission quality levels and adequate quality metrics, applicable in the recommendations of standardisation organisations, as well as suggested limit values conditioning acceptable voice data transmission quality were indicated and discussed. A packet network model was designed and tested, taking into account VoIP architecture supporting various audio codecs used for voice compression. Transmission mechanisms based on audio codecs G.711, G.723, G.726, G.728 and G.729 were investigated. It was shown that for delay-sensitive traffic which fluctuates beyond its nominal rate, selected codecs have an advantage over others and allow for better transmission quality of VoIP traffic with guaranteed bandwidth and delay.
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Bibliografia

[1] S. K. Puspita FM and S. Z. Taib BM, “Improved models of internet charging scheme of single bottleneck link in multi qos networks,” 2013. [Online]. Available: http://ddms.usim.edu.my:80/jspui/handle/123456789/15429
[2] A. R. Modarressi and S. Mohan, “Control and management in next-generation networks: challenges and opportunities,” IEEE Communications Magazine, vol. 38, no. 10, pp. 94–102, 2000. [Online]. Available: https://doi.org/10.1109/35.874976
[3] D. Strzęciwilk, K. Ptaszek, P. Hoser, and I. Antoniku, “A research on the impact of encryption algorithms on the quality of vpn tunnels’ transmission,” in ITM Web of Conferences, vol. 21. EDP Sciences, 2018, p. 00011. [Online]. Available: https://doi.org/10.1051/itmconf/ 20182100011
[4] H. J. Kim and S. G. Choi, “A study on a qos/qoe correlation model for qoe evaluation on iptv service,” in 2010 The 12th International Conference on Advanced Communication Technology (ICACT), vol. 2. IEEE, 2010, pp. 1377–1382.
[5] D. Strzęciwilk, “Examination of transmission quality in the ip multiprotocol label switching corporate networks,” International Journal of Electronics and Telecommunications, vol. 58, pp. 267–272, 2012. [Online]. Available: http://doi.org/10.2478/v10177-012-0037-z
[6] A. J. Estepa, R. Estepa, J. M. Vozmediano, and P. Carrillo, “Dynamic voip codec selection on smartphones,” Netw. Protoc. Algorithms, vol. 6, no. 2, pp. 22–37, 2014. [Online]. Available: https://doi.org/10.5296/npa.v6i2.5370
[7] W. M. Zuberek and D. Strzeciwilk, “Modeling traffic shaping and traffic policing in packet-switched networks,” Journal of Computer Sciences and Applications, vol. 6, no. 2, pp. 75–81, 2018. [Online]. Available: http://pubs.sciepub.com/jcsa/6/2/4
[8] D. Cohen, “Specifications for the network voice protocol,” UNIVERSITY OF SOUTHERN CALIFORNIA MARINA DEL REY INFORMATION SCIENCES INST, Tech. Rep., 1976. [Online]. Available: https://www.rfc-editor.org/info/rfc741
[9] J. Davidson, J. Peters, J. Peters, and B. Gracely, Voice over IP fundamentals. Cisco press, 2000. [10] S. Ganguly and S. Bhatnagar, VoIP: wireless, P2P and new enterprise voice over IP. John Wiley & Sons, 2008.
[11] B. Hartpence, Packet Guide to Voice over IP: A system administrator’s guide to VoIP technologies. " O’Reilly Media, Inc.", 2013.
[12] S. Deering and R. Hinden, “Rfc2460: Internet protocol, version 6 (ipv6) specification,” 1998.
[13] K. Ramakrishnan, S. Floyd, and D. Black, “Rfc3168: The addition of explicit congestion notification (ecn) to ip,” 2001.
[14] K. Nicholas, “Definition of the differentiated services field in the ipv4 and ipv6 headers,” RFC 2474, 1998.
[15] F. Baker, J. Polk, and M. Dolly, “A differentiated services code point (dscp) for capacity-admitted traffic,” Internet Engineering Task Force (IETF), 2010.
[16] D. Strzęciwilk, R. Nafkha, and R. Zawi´slak, “Performance analysis of a qos system with wfq queuing using temporal petri nets,” in International Conference on Computer Information Systems and Industrial Management. Springer, 2021, pp. 462–476. [Online]. Available: https://doi.org/10.1007/978-3-030-84340-3_38 [17] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, and W. Weiss, “An architecture for differentiated services,” 1998.
[18] D. C. Dowden, R. D. Gitlin, and R. L. Martin, “Next-generation networks,” Bell Labs technical journal, vol. 3, no. 4, pp. 3–14, 1998. [Online]. Available: https://doi.org/10.1002/bltj.2125
[19] G. R. Ash, Traffic engineering and QoS optimization of integrated voice and data networks. Elsevier, 2006.
[20] M. H. Miraz, S. A. Molvi, M. A. Ganie, M. Ali, and A. H. Hussein, “Simulation and analysis of quality of service (qos) parameters of voice over ip (voip) traffic through heterogeneous networks,” arXiv preprint arXiv:1708.01572, 2017. [Online]. Available: https://arxiv.org/abs/1708.01572
[21] E. T. Affonso, R. D. Nunes, R. L. Rosa, G. F. Pivaro, and D. Z. Rodriguez, “Speech quality assessment in wireless voip communication using deep belief network,” IEEE Access, vol. 6, pp. 77 022–77 032, 2018. [Online]. Available: https://doi.org/10.1109/ACCESS.2018.2871072
[22] J. Yu and I. Al-Ajarmeh, “Call admission control and traffic engineering of voip,” in 2007 Second International Conference on Digital Telecommunications (ICDT’07). IEEE, 2007, pp. 11–11.
[23] T. ITU, “Recommendation g. 114, one-way transmission time,” Series G: Transmission Systems and Media, Digital Systems and Networks, Telecommunication Standardization Sector of ITU, 2000.
[24] J. H. James, B. Chen, and L. Garrison, “Implementing voip: a voice transmission performance progress report,” IEEE Communications Magazine, vol. 42, no. 7, pp. 36–41, 2004. [Online]. Available: https://doi.org/10.1109/MCOM.2004.1316528
[25] J. G. Beerends, C. Schmidmer, J. Berger, M. Obermann, R. Ullmann, J. Pomy, and M. Keyhl, “Perceptual objective listening quality assessment (polqa), the third generation itut standard for end-to-end speech quality measurement part i—temporal alignment,” Journal of the Audio Engineering Society, vol. 61, no. 6, pp. 366–384, 2013. [Online]. Available: http://resolver.tudelft.nl/uuid:91d98cbc-d802-40d3-a1bb-a58d67668728
[26] R. D. Nunes, R. L. Rosa, and D. Z. Rodríguez, “Performance improvement of a non-intrusive voice quality metric in lossy networks,” IET Communications, vol. 13, no. 20, pp. 3401–3408, 2019. [Online]. Available: https://doi.org/10.1049/iet-com.2018.5165
[27] B. Naderi and R. Cutler, “An open source implementation of itu-t recommendation p. 808 with validation,” arXiv preprint arXiv:2005.08138, 2020. [Online]. Available: https://arxiv.org/ct?url=https%3A%2F%2Fdx. doi.org%2F10.21437%2FInterspeech.2020-2665&v=69f1738e
[28] A. W. Rix, J. G. Beerends, M. P. Hollier, and A. P. Hekstra, “Perceptual evaluation of speech quality (pesq)-a new method for speech quality assessment of telephone networks and codecs,” in 2001 IEEE international conference on acoustics, speech, and signal processing. Proceedings (Cat. No. 01CH37221), vol. 2. IEEE, 2001, pp. 749–752.
[29] S. Voran, “Objective estimation of perceived speech quality. i. development of the measuring normalizing block technique,” IEEE Transactions on speech and audio processing, vol. 7, no. 4, pp. 371–382, 1999. [Online]. Available: https://doi.org/10.1109/89.771259
[30] M. Coto-Jimenez, J. Goddard-Close, L. Di Persia, and H. L. Rufiner, “Hybrid speech enhancement with wiener filters and deep lstm denoising autoencoders,” in 2018 IEEE International Work Conference on Bioinspired Intelligence (IWOBI). IEEE, 2018, pp. 1–8. [Online]. Available: https://doi.org/10.1109/IWOBI.2018.8464132
[31] L. Ding and R. A. Goubran, “Speech quality prediction in voip using the extended e-model,” in GLOBECOM’03. IEEE Global Telecommunications Conference (IEEE Cat. No. 03CH37489), vol. 7. IEEE, 2003, pp. 3974–3978. [Online]. Available: https://doi.org/10.1109/GLOCOM.2003.1258975
[32] J. A. Bergstra and C. Middelburg, “Itu-t recommendation g. 107: The e-model, a computational model for use in transmission planning,” 2003.
[33] R. Jain, “Quality of experience,” IEEE multimedia, vol. 11, no. 1, pp. 96–95, 2004. [Online]. Available: https://doi.org/10.1109/MMUL.2004.10000
[34] A. Eskandar, M. Syed et al., “Performance analysis of voip over gre tunnel.” International Journal of Computer Network & Information Security, vol. 7, no. 12, 2015. [Online]. Available: http://doi.org/10.5815/ijcnis.2015.12.01
[35] R. S. Ramakrishnan and P. V. Kumar, “Performance analysis of different codecs in voip using sip,” in The Conference on Mobile and Pervasive Computing, 2008, pp. 142–145.
[36] S. Ragot, B. Kovesi, R. Trilling, D. Virette, N. Duc, D. Massaloux, S. Proust, B. Geiser, M. Gartner, S. Schandl et al., “Itu-t g. 729.1: An 8-32 kbit/s scalable coder interoperable with g. 729 for wideband telephony and voice over ip,” in 2007 IEEE International Conference on Acoustics, Speech and Signal Processing-ICASSP’07, vol. 4. IEEE, 2007, pp. IV–529. [Online]. Available: https://doi.org/10.1109/ICASSP. 2007.366966
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Autorzy i Afiliacje

Dariusz Strzęciwilk
1

  1. Institute of Information Technology, University of Life Sciences, Warsaw, Poland

Abstrakt

We present the implementation of the hardware ANS compressor in FPGAs. The main goal of the design was to propose a solution suitable to low-cost, low-energy embedded systems. We propose the streaming-rANS algorithm of the ANS family as a target for the implementation. Also, we propose a set of algorithm parameters that substantially reduce the use of FPGA resources, and we examine what is the influence of the chosen parameters on compression performance. Further, we compare our design to the lossless codecs found in literature, and to the streaming-rANS codecs with arbitrary parameters.
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Autorzy i Afiliacje

Magdalena Pastuła
1
Paweł Russek
1
Kazimierz Wiatr
1

  1. AGH University of Krakow, Krakow, Poland

Abstrakt

Since speaker recognition and verification became heavily used technology, both in professional applications like forensics and more everyday ones, the question arose: what factors can impact results of those processes? One thing that may be important with respect to this subject is lossy coding, as some of the information contained in an original file is lost in the coding process. In the era of globalization, not only native languages or languages of neighboring countries are of interest to researchers, but also those quite far, especially from Asia – the biggest exporter of goods and services to Europe. Those economic relationships are usually connected with the interchange of personnel, which further shortens geographical distance. The article presents the results that are a continuation of research on the behavior of Japanese language formants. Earlier research focused on changes occurring for the first and second formants. This article presents changes observed for the third and fourth formants. The knowledge of these changes is indicated in the process of speaker identification in forensics using the spectrographic method. At the Department of Acoustics and Multimedia, Wroclaw University of Science and Technology and in many centers around the world, the auditoryspectrographic method is used, which is a combination of the aural and spectrographic methods. In the spectrographic part, a person is identified on the basis of a comparison of the formants’ trajectory.
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Autorzy i Afiliacje

Mateusz Andrzej Kucharski
1
Stefan Brachmański
1
ORCID: ORCID

  1. Wroclaw University of Science and Technology, Wroclaw, Poland

Abstrakt

The MDCT and IntMDCT Algorithm is widely utilized is Audio coding. By lifting scheme or rounding operation IntegerMDCT is evolved from Modified Discrete Cosine Transform. This method acquire the properties of MDCT and contribute excelling invertiblity and good spectral mean .In this paper we discuss about the audio codec like AAC and FLAC using MDCT and Integer MDCT algorithm and to find which algorithm shows better Compression Ratio(CR).The confines of this task is to hybriding lossy and lossless audio codec with diminished bit rate but with finer sound quality. Certainly the quality of the audio is figure out by Subjective and Objective testing which is in terms of MOS (Mean opinion square), ABx and some of the hearing aid testing methodology like PEAQ(Perceptual Evaluation Audio Quality) and ODG(Objective Difference Grade)is followed. Execution measure, that is Compression Ratio(CR) and Sound Pressure Level (SPL) is approximated.

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Autorzy i Afiliacje

M. Davidson Kamala Dhas
R. Priyadharsini

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