Format-independent media delivery, applied to RTP, MP4, and Ogg

The current multimedia landscape is characterized by a significant heterogeneity in terms of coding and delivery formats, usage environments, and user preferences. This paper introduces a transparent multimedia content adaptation and delivery approach, i.e., model-driven content adaptation and delivery. It is based on a model that takes into account the structural metadata, semantic metadata, and scalability information of media bitstreams. Further, a format-independent multimedia packaging method is proposed based on this model for media bitstreams and MPEG-B BSDL. Thus, multimedia packaging is obtained by encapsulating the selected and adapted structural metadata within a specific delivery format. This packaging process is implemented using XML transformation filters and MPEG-B BSDL. To illustrate this format-independent packaging technique, we apply it to three packaging formats: RTP, MP4, and Ogg

Network Performance in HTML5 Video Connections

[EN] Currently, most of remote education systems use video streaming as the main basis to support teaching. These emissions can be seen in devices with different hardware features such as personal computers, tablets or smartphones through networks with different capacities. The use of different web browsers and coding options can also influence the network performance. Therefore, the quality of the video displayed may be different. This work presents a practical study to establish the best combination of web browsers and containers to encode multimedia files for videos streaming in personal computers running Windows 7 and Windows 10 operating systems. For this, a video encoded with different codecs and compressed with different containers have been transmitted through a 1000BaseT network. Finally, the results are analyzed and compared to determine which would be the most efficient combination of parameters according to the resolution of the transmitted video.This work has been partially supported by the European Union through the ERANETMED (Euromediterranean Cooperation through ERANET joint activities and beyond) project ERANETMED3-227 SMARTWATIR and by the Ministerio de Educación, Cultura y Deporte , through the Convocatoria 2016 - Proyectos I+D+I - Programa Estatal De Investigación, Desarrollo e Innovación Orientada a los retos de la sociedad (Project TEC2016-76795-C6-4-R) and through the Convocatoria 2017 - Proyectos I+D+I - Programa Estatal de Investigación, Desarrollo e Innovación, convocatoria excelencia (Project TIN2017-84802-C2-1-P).Sendra, S.; Túnez-Murcia, AI.; Lloret, J.; Jimenez, JM. (2018). Network Performance in HTML5 Video Connections. Network Protocols and Algorithms. 10(3):43-62. https://doi.org/10.5296/npa.v10i3.13933S436210

Database of audio records

Diplomka a prakticky castDiplome with partical part

Streaming of compressed multimedia content

Tato diplomová práce pojednává o přenosu multimédií po síti za pomoci komprimačních algoritmů užívaných v kodeku MPEG-4 part 10 a Real Time Protokolu (RTP). Čtenář zde bude seznámen se základními pojmy ohledně multimédií. Bude pojednáno o principech komprimačních technik pro audio i video a bude odůvodněna nutnost použití komprimace při přenosu multimediálních dat po síti. Rovněž budou popsány principy a vlastnosti protokolu RTP, jež je pro stream audia a videa navržen. Budou vybrány vhodné kodeky (Theora pro video a Vorbis pro audio) pro implementaci dvou aplikací (server, klient), jež mají za úkol načíst surová data, zkomprimovat je pomocí kodeku Theora (Vorbis), přenést po síti v podobě RTP paketů, zachytit klientem a reprodukovat koncovému uživateli. Samozřejmostí je kompletní popis implementace obou aplikací napsaných v jazyce C/C++ a Java. Čtenáři by si měl z práce odnést základní přehled o celé problematice a načerpat zkušenosti s implementací, jež jsou zde shrnuty.The topic of the master's thesis is about transmission of multimedia via network with help of compression’s algorithm used in codec MPEG-4 part 10 and Real Time Protocol (RTP). First part of this master thesis will be familiarization with basic terms about multimedia. It will be queried about techniques of compression axioms for audio and video, justification for necessary use of compression in transmitting multimedia via network. Also, here will be subscribed principles and properties of Real Time Protocol, which was designed for multimedia stream. From the start, competent codec for implementation of two applications (client, server) with read raw video and audio data, will choose the following : compression with help of codec Theora (Vorbis), transfer via network in RTP packets, decode and play at visual form for end user. Complete subscription of implementation for both applications written in programming languages C/C++ and Java, will be certain. The basic overview about this problematic experience with implementation, which were here concluded, are the reasons for reading this master thesis.

Compression, decompression and streaming of video data

Zaměřením této bakalářské je popsání schopností nástroje FFmpeg a kodeku Theora. Porovnání nástrojů a technologií kodeků MPEG-4 (MPEG-4 ASP, H.264), Theora a MPEG-2. Testování komprimačních schopností a srovnání výstupních snímků videa komprimovaném v MPEG4-AVC kodeku, Theoře a videa bez komprese. Ukázka postupu a použití programovacího rozhraní Theora a FFmpeg pro práci s multimediálním obsahem. Částí práce je také analýza a implementace aplikace přehrávače vytvořeného s pomocí nástroje FFmpeg a aplikace pro přenos Theora video paketů pomocí RTP protokolu.The Bachelor thesis is focused on description of FFmpeg tool and theora codec abilities. Comparison of capabilities and technologies within MPEG-4 (MPEG4- ASP, H.264), Theora and MPEG-2 codec. Testing the compression efficiency and output image quality of codecs x264 and Theora. Further I mentioned samples of using Theora and FFmpeg programming interface for video handling. Part of the thesis is also analysis and implementation of the player based on FFmpeg libraries and application for streaming Theora encoded video packets by RTP protocol.

Compare multimedia frameworks in mobile platforms

Multimedia feature is currently one of the most important features in mobile devices. Many modern mobile platforms use a centralized software stack to handle multimedia requirements that software stack is called multimedia framework. Multimedia framework belongs to the middleware layer of mobile operating system. It can be considered as a bridge that connects mobile operating system kernel, hardware drivers with UI applications. It supplies high level APIs that offers simple and easy solutions for complicated multimedia tasks to UI application developers. Multimedia Framework also manages and utilizes low lever system software and hardware in an efficient manner. It offers a centralize solution between high level demands and low level system resources. In this M.Sc. thesis project we have studied, analyzed and compared open source GStreamer, Android Stagefright and Microsoft Silverlight Media Framework from several perspectives. Some of the comparison perspectives are architecture, supported use cases, extensibility, implementation language and program language support (bindings), developer support, and legal status aspects. One of the main contributions of this thesis work is that clarifying in details the strength and weaknesses of each framework. Furthermore, the thesis should serve decision-making guidance when on needs to select a multimedia framework for a project. Moreover, and to enhance the impression with the three multimedia frameworks, a basic media player implementation is demonstrated with source code in the thesis.fi=Opinnäytetyö kokotekstinä PDF-muodossa.|en=Thesis fulltext in PDF format.|sv=Lärdomsprov tillgängligt som fulltext i PDF-format

Test bed design for interactive video conference services

In the last decade, telecommunication industry has been widely developed in area of the multimedia interactive services. Actually, there are still emerging new videoconference and instant messaging applications. For the provisioning of multimedia interactive services it is essential to provide a required level of customer satisfaction, given by the end-user quality. The video and audio compression improvement of the newest video coding standards H.264/AVC and AAC allows for providing video and audio streaming for low bit and frame rates while preserving the perceptual quality. This is especially suitable for interactive multimedia applications in 3G wireless networks. The aim of this thesis was to design the ”State of the Art” video conferencing environment supporting H.264/AVC and AAC codecs. Moreover, this environment provides opportunity to analyze end user quality at all layer of OSI model. As a result of this design, we have an open source application, that offers a good quality of image and sound at very low rates (90kbps, 9fps for video) at the same time that reduces the reception delay that now exists in commercial applications

Opus audiokoodekki matkapuhelinverkoissa

The latest generations in mobile networks have enabled a possibility to include high quality audio coding in data transmission. On the other hand, an on-going effort to move the audio signal processing from dedicated hardware to data centers with generalized hardware introduces a challenge of providing enough computational power needed by the virtualized network elements. This thesis evaluates the usage of a modern hybrid audio codec called Opus in a virtualized network element. It is performed by integrating the codec, testing it for functionality and performance on a general purpose processor, as well as evaluating the performance in comparison to the digital signal processor's performance. Functional testing showed that the codec was integrated successfully and bit compliance with the Opus standard was met. The performance results showed that although the digital signal processor computes the encoder's algorithms with less clock cycles, related to the processor's whole capacity the general purpose processor performs more efficiently due to higher clock frequency. For the decoder this was even clearer, when the generic hardware spends on average less clock cycles for performing the algorithms.Uusimmat sukupolvet matkapuhelinverkoissa mahdollistavat korkealaatuisen audiokoodauksen tiedonsiirrossa. Toisaalta audiosignaalinkäsittelyn siirtäminen sovelluskohtaisesta laitteistosta keskitettyjen palvelinkeskusten yleiskäyttöiseen laitteistoon on käynnissä, mikä aiheuttaa haasteita tarjota riittävästi laskennallista tehoa virtualisoituja verkkoelementtejä varten. Tämä diplomityö arvioi modernin hybridikoodekin, Opuksen, käyttöä virtualisoidussa verkkoelementissä. Se on toteutettu integroimalla koodekki, testaamalla funktionaalisuutta ja suorituskykyä yleiskäyttöisellä prosessorilla sekä arvioimalla suorituskykyä verrattuna digitaalisen signaaliprosessorin suorituskykyyn. Funktionaalinen testaus osoitti että koodekki oli integroitu onnistuneesti ja että bittitason yhdenmukaisuus Opuksen standardin kanssa saavutettiin. Suorituskyvyn testitulokset osoittivat, että vaikka enkoodaus tuotti vähemmän kellojaksoja digitaalisella signaaliprosessorilla, yleiskäyttöinen prosessori suoriutuu tehokkaammin suhteutettuna prosessorin kokonaiskapasiteettiin korkeamman kellotaajuuden ansiosta. Dekooderilla tämä näkyi vielä selkeämmin, sillä yleiskäyttöinen prosessori kulutti keskimäärin vähemmän kellojaksoja algoritmien suorittamiseen

Linux-Box: DVB and VoD streaming over local area networks

Aquest treball tracta sobre un projecte comú anomenat Linux-Box portat a terme per diferents persones al departament de Telecomunicacions (IET) de la Universitat de Pisa. Linux-Box és un sistema dotat amb targetes TDT (DVB-T) i de televisió per satèl·lit (DVB-S) que permet transmetre aquests senyals fins a un àmbit domèstic. Més endavant podria ser utilitzat en àmbits privats com les cases de clients o en institucions públiques com escoles, universitats, biblioteques i també seria possible en àmbits empresarials. El projecte està dividit en 4 apartats: 1. Ubuntu 6.06 LTS. Explica perquè s’utilitza Ubuntu en el projecte. A més també s’explica de forma breu que és Linux i les distribucions més utilitzades. 2. Multimedia Network Protocols: s’expliquen els diferents protocols desde la capa de xarxa fins la capa d’aplicació que s’utilitzen en el projecte Linux-Box. Aquests protocols són utilitzats tant en streaming, com en anunciació, unicast/multicast, encapsulat de vídeo i codecs. Els diversos temes tractats aquí es fan amb el propòsit de comparar i no només com a recerca teòrica. A la fi es veuen els programes utilitzats en el projecte per analitzar el tràfic de la xarxa. 3. Linux-Box: s’explica el funcionament i els objectius globals del projecte. Es dedica un sub-apartat a “VideoLan - VLC” part important a nivell de sofware. Més endavant es parla de les característiques de la Linux-Box de forma acurada: streaming de VoD i senyals de TV i s’analitzen els problemes coneguts i les seves solucions proposades. A la fi s’enumeren els llenguatges de programació utilitzats al projecte i en quina part s’utilitzen. Observarem que és una aplicació on diversos llenguatges de programació estan contínuament solapats. 4. Developed Part: es posa en pràctica la teoria estudiada a la resta del treball. Està dividida en 4 seccions: Desenvolupar una aplicació en codi C per convertir la llista de Canals (tant terrestre com de satèl·lit) en format XML. Una secció dedicada al streaming de Canals de TV a la pàgina web principal. Un anàlisis profund dels paquets creats per la Linux-Box i la seva activitat a la xarxa. Finalment s’analitzen els diferents scripts i les seves configuracions. Alguns són útils per a un futur desenvolupament i d’altres s’utilitzen en seccions prèvies. 5. Conclusions: conté les conclusions i línies futures. El projecte compta amb diverses opcions que encara poden ser implementades i estudiades. Aquí exposem les nostres interpretacions i possibles línies futures d’estudi