Investigation of Different Video Compression Schemes Using Neural Networks

Image/Video compression has great significance in the communication of motion pictures and still images. The need for compression has resulted in the development of various techniques including transform coding, vector quantization and neural networks. this thesis neural network based methods are investigated to achieve good compression ratios while maintaining the image quality. Parts of this investigation include motion detection, and weight retraining. An adaptive technique is employed to improve the video frame quality for a given compression ratio by frequently updating the weights obtained from training. More specifically, weight retraining is performed only when the error exceeds a given threshold value. Image quality is measured objectively, using the peak signal-to-noise ratio versus performance measure. Results show the improved performance of the proposed architecture compared to existing approaches. The proposed method is implemented in MATLAB and the results obtained such as compression ratio versus signalto- noise ratio are presented

Concealment algorithms for networked video transmission systems

This thesis addresses the problem of cell loss when transmitting video data over an ATM network. Cell loss causes sections of an image to be lost or discarded in the interconnecting nodes between the transmitting and receiving locations. The method used to combat this problem is to use a technique called Error Concealment, where the lost sections of an image are replaced with approximations derived from the information in the surrounding areas to the error. This technique does not require any additional encoding, as used by Error Correction. Conventional techniques conceal from within the pixel domain, but require a large amount of processing (2N2 up to 20N2) where N is the dimension of an N×N square block. Also, previous work at Loughborough used Linear Interpolation in the transform domain, which required much less processing, to conceal the error. [Continues.

Flexi-WVSNP-DASH: A Wireless Video Sensor Network Platform for the Internet of Things

abstract: Video capture, storage, and distribution in wireless video sensor networks (WVSNs) critically depends on the resources of the nodes forming the sensor networks. In the era of big data, Internet of Things (IoT), and distributed demand and solutions, there is a need for multi-dimensional data to be part of the Sensor Network data that is easily accessible and consumable by humanity as well as machinery. Images and video are expected to become as ubiquitous as is the scalar data in traditional sensor networks. The inception of video-streaming over the Internet, heralded a relentless research for effective ways of distributing video in a scalable and cost effective way. There has been novel implementation attempts across several network layers. Due to the inherent complications of backward compatibility and need for standardization across network layers, there has been a refocused attention to address most of the video distribution over the application layer. As a result, a few video streaming solutions over the Hypertext Transfer Protocol (HTTP) have been proposed. Most notable are Apple’s HTTP Live Streaming (HLS) and the Motion Picture Experts Groups Dynamic Adaptive Streaming over HTTP (MPEG-DASH). These frameworks, do not address the typical and future WVSN use cases. A highly flexible Wireless Video Sensor Network Platform and compatible DASH (WVSNP-DASH) are introduced. The platform's goal is to usher video as a data element that can be integrated into traditional and non-Internet networks. A low cost, scalable node is built from the ground up to be fully compatible with the Internet of Things Machine to Machine (M2M) concept, as well as the ability to be easily re-targeted to new applications in a short time. Flexi-WVSNP design includes a multi-radio node, a middle-ware for sensor operation and communication, a cross platform client facing data retriever/player framework, scalable security as well as a cohesive but decoupled hardware and software design.Dissertation/ThesisDoctoral Dissertation Electrical Engineering 201

Secure high definition video conferencing

The aim of this review is to study technologies involved in a video conference through Internet. Some security issues and solutions to them are also covered in this report. At first, several video conference environments are presented to clarify concepts. Also some commercial solutions are mentioned. Secondly, signalling protocols, specially SIP, are studied to be used in the set up of a video conference. Possibilities to secure SIP are also covered in the theoretical study. Thirdly, the Secure RTP protocol is presented to be used to protect the media flows. Then a key agreement mechanism, MIKEY, is stated to make the key agreement needed to establish a crypto session for SRTP. After the background study, an implementation of a secure video conferencing platform using miniSIP and RTP Packet Reflector is proposed. Then, the final implementation is detailed, showing up the problems appeared during this process and possible solutions to them. Possible new features for the system are also proposed. Finally, some measurement results taken using the new software are presented and analyzed

Implementation of a MPEG 1 layer I audio decoder with variable bit lengths

One of the most popular forms of audio compression is MPEG (Moving Picture Experts Group). By using a VHDL (Very high-speed integrated circuit Hardware Description Language) implementation of a MPEG audio decoder and varying the word length of the constants and the multiplications used in the decoding process, and comparing the error, the minimum word length required can be determined. In general, the smaller the word length, the smaller the hardware resources required. This thesis is an investigation to find the minimum bit lengths required for each of the four multiplication sections used in a MPEG Audio decoder, that will still meet the quality levels specified in the MPEG standard. The use of the minimum bit lengths allows the minimum area resources of a FPGA (Field Programmable Gate Array) to be used. A FPGA model was designed that allowed the number of bits used to represent four constants and the results of the multiplications using these constants to vary. In order to limit the amount of data generated, testing was restricted to a single channel of audio data sampled at a frequency of 32kHz. This was then compared to the supplied C model distributed with the MPEG Audio Standard. It was found that for the MPEG audio coder to be fully compliant with the standard the bit lengths of the constants and the multiplications could be reduced by 75% and to be partial compliant with the standard, the bit lengths of the constants and the multiplications could be reduced by up to 82%. An implementation of a MPEG audio decoder in VHDL has the advantage of specific hardware, optimised, for all the different complex mathematical operations thereby reducing the repetitive operations and therefore power consumption and the time required performing these complex operations

Quality of Service Controlled Multimedia Transport Protocol

PhDThis research looks at the design of an open transport protocol that supports a range of services including multimedia over low data-rate networks. Low data-rate multimedia applications require a system that provides quality of service (QoS) assurance and flexibility. One promising field is the area of content-based coding. Content-based systems use an array of protocols to select the optimum set of coding algorithms. A content-based transport protocol integrates a content-based application to a transmission network. General transport protocols form a bottleneck in low data-rate multimedia communicationbsy limiting throughpuot r by not maintainingt iming requirementsT. his work presents an original model of a transport protocol that eliminates the bottleneck by introducing a flexible yet efficient algorithm that uses an open approach to flexibility and holistic architectureto promoteQ oS.T he flexibility andt ransparenccyo mesi n the form of a fixed syntaxt hat providesa seto f transportp rotocols emanticsT. he mediaQ oSi s maintained by defining a generic descriptor. Overall, the structure of the protocol is based on a single adaptablea lgorithm that supportsa pplication independencen, etwork independencea nd quality of service. The transportp rotocol was evaluatedth rougha set of assessmentos:f f-line; off-line for a specific application; and on-line for a specific application. Application contexts used MPEG-4 test material where the on-line assessmenuts eda modified MPEG-4 pl; yer. The performanceo f the QoSc ontrolledt ransportp rotocoli s often bettert hano thers chemews hen appropriateQ oS controlledm anagemenatl gorithmsa re selectedT. his is shownf irst for an off-line assessmenwt here the performancei s compared between the QoS controlled multiplexer,a n emulatedM PEG-4F lexMux multiplexers chemea, ndt he targetr equirements. The performanceis also shownt o be better in a real environmentw hen the QoS controlled multiplexeri s comparedw ith the real MPEG-4F lexMux scheme

Real Time Multiple Codecs Switching Architecture For Video Conferencing.

Trend terkini yang berhubung kait dengan khidmat video internet telah merubah dengan mendadak cara kita berfikir dan berkerja. Perkhidmatan-perkhidmatan ini telah memperluaskan ruang lingkup teknologi serta memberi peluang kepada perkembangan idea-idea penyiaran tanpa sempadan dan memendekkan jarak komunikasi secara geografi melalui telesidang video. Current trends related to internet video services have drastically transformed the way we think and work. These services have expanded the new scope of technology and have given opportunity for broadcasting ideas without any boundaries as well as shortened the vast geographical distances for communications through video conferencing

MediaSync: Handbook on Multimedia Synchronization

This book provides an approachable overview of the most recent advances in the fascinating field of media synchronization (mediasync), gathering contributions from the most representative and influential experts. Understanding the challenges of this field in the current multi-sensory, multi-device, and multi-protocol world is not an easy task. The book revisits the foundations of mediasync, including theoretical frameworks and models, highlights ongoing research efforts, like hybrid broadband broadcast (HBB) delivery and users' perception modeling (i.e., Quality of Experience or QoE), and paves the way for the future (e.g., towards the deployment of multi-sensory and ultra-realistic experiences). Although many advances around mediasync have been devised and deployed, this area of research is getting renewed attention to overcome remaining challenges in the next-generation (heterogeneous and ubiquitous) media ecosystem. Given the significant advances in this research area, its current relevance and the multiple disciplines it involves, the availability of a reference book on mediasync becomes necessary. This book fills the gap in this context. In particular, it addresses key aspects and reviews the most relevant contributions within the mediasync research space, from different perspectives. Mediasync: Handbook on Multimedia Synchronization is the perfect companion for scholars and practitioners that want to acquire strong knowledge about this research area, and also approach the challenges behind ensuring the best mediated experiences, by providing the adequate synchronization between the media elements that constitute these experiences

Analyzing Voice And Video Call Service Performance Over A Local Area Network

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2010Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2010Bu çalışmada, VOIP teknolojisinden ve bu teknolojiyi kablolu ve kablosuz ortamda gerçeklemenin en önemli darboğazları anlatılacaktır. Ayrıca H.323, SIP (Session Initiation Protocol), Megaco ve MGCP gibi yaygın olarak kullanılan ses iletim protokolleri ve H.261, H.263 ve H.264 gibi görüntü iletim protokollerinden bahsedilmiştir. Ses kodek seçimi ve VOIP servis kalitesine etki eden faktörleri anlatılmaktadır. Bu tezde, ses, görüntü ve veri iletişimini aynı anda bünyesinde barındıran gerçek şebekeler simüle edilecektir. Kullanıcılara rastlantısal olarak ses, görüntü ve FTP gibi birtakım uygulamalar atanmıştır. Ayrıca önerilen kablolu şebekeye, kablosuz bir şebeke ilave edilerek sonuçlar incelenecektir. Optimal servis kalitesini sağlamak için seçilen uygun kuyruklama mekanizmaları ve kodek seçimlerini içeren senaryolar incelenecek ve OPNET ile elde edilmiş simülasyon sonuçları tartışılacaktır.In this study, we present a detailed description of the VoIP and also the most common challenges of implementing voice communication into wireline or wireless networks are discussed. Common voice protocols, such as H.323, Session Initiation Protocol (SIP), Megaco, MGCP and video protocols such as H.261, H.263, H.264 are described as well. CODEC selection and factors affecting VoIP Quality of Service are analyzed. We simulate a real network which includes both voice, video and data communication simultaneously. Workstations are randomly assigned to different applications, such as voice, video and FTP. We will also implement a wireless network to our proposed system. The scenarios including selecting appropriate queuing scheme and codec selection are presented and the simulation results with OPNET are drawn.Yüksek LisansM.Sc

Media Processing in Video Conferences for Cooperating Over the Top and Operator Based Networks

Telecom operators have dominated the communication industry for a long time by providing services with guaranteed quality of service. Such services are provided by the operator at the cost of maintaining a high grade network. With the introduction of broadband and internet, many over the top (OTT) services have emerged. These services use the underlying operator networks as a mere bit pipe while all service intelligence resides in the application running on the client device. Introduction of OTT services has seen a good response from general users who are no longer bound to services provided by the network operator. This in turn has caused operators and telecom companies to loose the ownership of their customers. This thesis takes media processing in video conferencing as a case study to compare the two competing domains of operator networks and OTT networks. Both domains offer video conferencing to end users, but they follow different architectures. The study shows that OTT services can perform much better if they utilize support of the underlying network. This will also bring the user base back to the network operator. The proposal is to turn the competition into cooperation between both parties. Assessments are done from both technical as well as business perspectives to assert that such cooperative agreements are possible and should be experimented in real life