Multi-core platforms for audio and multimedia coding algorithms in telecommunications

Tietoliikenteessä käytettävät multimedian koodausalgoritmit eli koodekit kehittyvät jatkuvasti. USAC ja Opus ovat uusia, sekä puheelle että musiikille soveltuvia audiokoodekkeja. Molemmat ovat sijoittuneet korkealle koodekkien äänenlaatua vertailevissa tutkimuksissa. Näiden keskeisiä ominaisuuksia käsitellään kirjallisuuskatsaukseen perustuen. Varsinkin HD-tasoisen videon käsittelyssä käytettävät koodekit vaativat suurta laskentatehoa. Tilera TILEPro64 -moniydinsuorittimen ja sille optimoitujen multimediakoodekkien suorituskykyä testattiin tarkoitukseen suunnitelluilla tietokoneohjelmilla. Tulokset osoittivat, että suoritinytimiä lisättäessä videon koodausalgoritmien suoritusnopeus kasvaa tiettyyn rajaan asti. Testatuilla äänen koodausalgoritmeillä ytimien lisääminen ei parantanut suoritusnopeutta. Tileran moniydinratkaisuja verrattiin lopuksi Freescalen ja Texas Instrumentsin moniydinratkaisuihin. Huolimatta eroista laitteistoarkkitehtuureissa, kyseisten toimittajien kehitystyökaluissa todettiin olevan paljon samoja piirteitä.Multimedia coding algorithms used in telecommunications evolve constantly. Benefits and properties of two new hybrid audio codecs (USAC, Opus) were reviewed on a high level as a literature study. It was found that both have succeeded well in subjective sound quality measurements. Tilera TILEPro64-multicore platform and a related software library was evaluated in terms of performance in multimedia coding. The performance in video coding was found to increase with the number of processing cores up to a certain point. With the tested audio codecs, increasing the number of cores did not increase coding performance. Additionally, multicore products of Tilera, Texas Instruments and Freescale were compared. Development tools of all three vendors were found to have similar features, despite the differences in hardware architectures

Network streaming and compression for mixed reality tele-immersion

Bulterman, D.C.A. [Promotor]Cesar, P.S. [Copromotor

VoipLoc : VoIP call provenance using acoustic side-channels

We develop a novel technique to determine call provenance in anonymous VoIP communications using acoustic side-channels. The technique exploits location-attributable information embedded within audio speech data. The victim’s speech is exploited as an excitation signal, which is modulated (acted upon) by the acoustic reflection characteristics of the victim’s location. We show that leading VoIP communication channels faithfully transfer this information between sender-receiver pairs, enabling passive receivers to extract a location fingerprint, to establish call provenance. To establish provenance, a fingerprint is compared against a database of labelled fingerprints to identify a match. The technique is fully passive and does not depend on any characteristic background sounds, is speaker independent, and is robust to lossy network conditions. Evaluation using a corpus of recordings of VoIP conversations, over the Tor network, confirms that recording locations can be fingerprinted and detected remotely with low false-positive rate

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

Recent Advances in Steganography

Steganography is the art and science of communicating which hides the existence of the communication. Steganographic technologies are an important part of the future of Internet security and privacy on open systems such as the Internet. This book's focus is on a relatively new field of study in Steganography and it takes a look at this technology by introducing the readers various concepts of Steganography and Steganalysis. The book has a brief history of steganography and it surveys steganalysis methods considering their modeling techniques. Some new steganography techniques for hiding secret data in images are presented. Furthermore, steganography in speeches is reviewed, and a new approach for hiding data in speeches is introduced

Secure covert communications over streaming media using dynamic steganography

Streaming technologies such as VoIP are widely embedded into commercial and industrial applications, so it is imperative to address data security issues before the problems get really serious. This thesis describes a theoretical and experimental investigation of secure covert communications over streaming media using dynamic steganography. A covert VoIP communications system was developed in C++ to enable the implementation of the work being carried out. A new information theoretical model of secure covert communications over streaming media was constructed to depict the security scenarios in streaming media-based steganographic systems with passive attacks. The model involves a stochastic process that models an information source for covert VoIP communications and the theory of hypothesis testing that analyses the adversary‘s detection performance. The potential of hardware-based true random key generation and chaotic interval selection for innovative applications in covert VoIP communications was explored. Using the read time stamp counter of CPU as an entropy source was designed to generate true random numbers as secret keys for streaming media steganography. A novel interval selection algorithm was devised to choose randomly data embedding locations in VoIP streams using random sequences generated from achaotic process. A dynamic key updating and transmission based steganographic algorithm that includes a one-way cryptographical accumulator integrated into dynamic key exchange for covert VoIP communications, was devised to provide secure key exchange for covert communications over streaming media. The discrete logarithm problem in mathematics and steganalysis using t-test revealed the algorithm has the advantage of being the most solid method of key distribution over a public channel. The effectiveness of the new steganographic algorithm for covert communications over streaming media was examined by means of security analysis, steganalysis using non parameter Mann-Whitney-Wilcoxon statistical testing, and performance and robustness measurements. The algorithm achieved the average data embedding rate of 800 bps, comparable to other related algorithms. The results indicated that the algorithm has no or little impact on real-time VoIP communications in terms of speech quality (< 5% change in PESQ with hidden data), signal distortion (6% change in SNR after steganography) and imperceptibility, and it is more secure and effective in addressing the security problems than other related algorithms

An Internet based multimedia infrastructure for collaborative engineering

Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Civil and Environmental Engineering, 2000.Includes bibliographical references (leaves 129-131).The evolution of computer based collaborative environments has resulted in easier and more economical design efforts among geographically distributed design teams. Most of today's internet based collaborative applications allow people that are geographically dispersed to meet with each other using their computers and work together without actually having to travel. A prototype system was developed by taking two tactical planning applications and incorporating them into the collaboration model employed by CAIRO (Collaborative Agent Interaction control and synchROnization). This system was developed based on the collaboration infrastructure that was developed as a part of the Da-Vinci Society Initiative at MIT. The main focus of this research lies in the formalization of a multi-media based architecture that supplements the existing collaboration infrastructure. This architecture lays the groundwork for development of a robust collaboration system that incorporates audio/video conferencing, speech recognition and synthesis and three-dimensional virtual meeting environments in order to facilitate efficient collaboration.by Padmanabha N. Vedam.S.M

Low-complexity video coding for receiver-driven layered multicast

In recent years, the “Internet Multicast Backbone,” or MBone, has risen from a small, research curiosity to a large- scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. In this paper, we describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an effi- cient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Inter- net). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a “real” application—the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone

Multimedia

The nowadays ubiquitous and effortless digital data capture and processing capabilities offered by the majority of devices, lead to an unprecedented penetration of multimedia content in our everyday life. To make the most of this phenomenon, the rapidly increasing volume and usage of digitised content requires constant re-evaluation and adaptation of multimedia methodologies, in order to meet the relentless change of requirements from both the user and system perspectives. Advances in Multimedia provides readers with an overview of the ever-growing field of multimedia by bringing together various research studies and surveys from different subfields that point out such important aspects. Some of the main topics that this book deals with include: multimedia management in peer-to-peer structures & wireless networks, security characteristics in multimedia, semantic gap bridging for multimedia content and novel multimedia applications

A Parametric Sound Object Model for Sound Texture Synthesis

This thesis deals with the analysis and synthesis of sound textures based on parametric sound objects. An overview is provided about the acoustic and perceptual principles of textural acoustic scenes, and technical challenges for analysis and synthesis are considered. Four essential processing steps for sound texture analysis are identifi ed, and existing sound texture systems are reviewed, using the four-step model as a guideline. A theoretical framework for analysis and synthesis is proposed. A parametric sound object synthesis (PSOS) model is introduced, which is able to describe individual recorded sounds through a fi xed set of parameters. The model, which applies to harmonic and noisy sounds, is an extension of spectral modeling and uses spline curves to approximate spectral envelopes, as well as the evolution of parameters over time. In contrast to standard spectral modeling techniques, this representation uses the concept of objects instead of concatenated frames, and it provides a direct mapping between sounds of diff erent length. Methods for automatic and manual conversion are shown. An evaluation is presented in which the ability of the model to encode a wide range of di fferent sounds has been examined. Although there are aspects of sounds that the model cannot accurately capture, such as polyphony and certain types of fast modulation, the results indicate that high quality synthesis can be achieved for many different acoustic phenomena, including instruments and animal vocalizations. In contrast to many other forms of sound encoding, the parametric model facilitates various techniques of machine learning and intelligent processing, including sound clustering and principal component analysis. Strengths and weaknesses of the proposed method are reviewed, and possibilities for future development are discussed