Steganalysis of ±k Steganography based on Noncausal Linear Predictor

The paper proposes a novel steganalytic technique for ±k steganographybased on noncausal linear predictor using prediction coefficients obtained from the autocorrelationmatrix for a block of pixels in the stego-image. The image is divided intoequal-size blocks, autocorrelation matrix is found for the block, and the appropriatenoncausal linear prediction coefficients is selected to predict all pixels in that block. Apixel is assumed to be embedded with message bit if the absolute difference betweenthe original pixel value and predicted pixel value exceeds the pre-defined threshold.The effectiveness of the proposed technique is verified using different images

Video compression with complete information for pre-recorded sources

Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2000.Includes bibliographical references (p. 123-130).by David Michael Baylon.Ph.D

DC coefficient restoration for transform image coding.

by Tse, Fu Wing.Thesis (M.Phil.)--Chinese University of Hong Kong, 1996.Includes bibliographical references (leaves 155-[63]).Acknowledgment --- p.iiiAbstract --- p.ivContents --- p.viList of Tables --- p.xList of Figures --- p.xiiNotations --- p.xviiChapter 1 --- Introduction --- p.1Chapter 1.1 --- DC coefficient restoration --- p.1Chapter 1.2 --- Model based image compression --- p.5Chapter 1.3 --- The minimum edge difference criterion and the existing estima- tion schemes --- p.7Chapter 1.3.1 --- Fundamental definitions --- p.8Chapter 1.3.2 --- The minimum edge difference criterion --- p.9Chapter 1.3.3 --- The existing estimation schemes --- p.10Chapter 1.4 --- Thesis outline --- p.14Chapter 2 --- A mathematical description of the DC coefficient restoration problem --- p.17Chapter 2.1 --- Introduction --- p.17Chapter 2.2 --- Mathematical formulation --- p.18Chapter 2.3 --- Properties of H --- p.22Chapter 2.4 --- Analysis of the DC coefficient restoration problem --- p.22Chapter 2.5 --- The MED criterion as an image model --- p.25Chapter 2.6 --- Summary --- p.27Chapter 3 --- The global estimation scheme --- p.29Chapter 3.1 --- Introduction --- p.29Chapter 3.2 --- the global estimation scheme --- p.30Chapter 3.3 --- Theory of successive over-relaxation --- p.34Chapter 3.3.1 --- Introduction --- p.34Chapter 3.3.2 --- Gauss-Seidel iteration --- p.35Chapter 3.3.3 --- Theory of successive over-relaxation --- p.38Chapter 3.3.4 --- Estimation of optimal relaxation parameter --- p.41Chapter 3.4 --- Using successive over-relaxation in the global estimation scheme --- p.43Chapter 3.5 --- Experiments --- p.48Chapter 3.6 --- Summary --- p.49Chapter 4 --- The block selection scheme --- p.52Chapter 4.1 --- Introduction --- p.52Chapter 4.2 --- Failure of the minimum edge difference criterion --- p.53Chapter 4.3 --- The block selection scheme --- p.55Chapter 4.4 --- Using successive over-relaxation with the block selection scheme --- p.57Chapter 4.5 --- Practical considerations --- p.58Chapter 4.6 --- Experiments --- p.60Chapter 4.7 --- Summary --- p.61Chapter 5 --- The edge selection scheme --- p.65Chapter 5.1 --- Introduction --- p.65Chapter 5.2 --- Edge information and the MED criterion --- p.66Chapter 5.3 --- Mathematical formulation --- p.70Chapter 5.4 --- Practical Considerations --- p.74Chapter 5.5 --- Experiments --- p.76Chapter 5.6 --- Discussion of edge selection scheme --- p.78Chapter 5.7 --- Summary --- p.79Chapter 6 --- Performance Analysis --- p.81Chapter 6.1 --- Introduction --- p.81Chapter 6.2 --- Mathematical derivations --- p.82Chapter 6.3 --- Simulation results --- p.92Chapter 6.4 --- Summary --- p.96Chapter 7 --- The DC coefficient restoration scheme with baseline JPEG --- p.97Chapter 7.1 --- Introduction --- p.97Chapter 7.2 --- General specifications --- p.97Chapter 7.3 --- Simulation results --- p.101Chapter 7.3.1 --- The global estimation scheme with the block selection scheme --- p.101Chapter 7.3.2 --- The global estimation scheme with the edge selection scheme --- p.113Chapter 7.3.3 --- Performance comparison at the same bit rate --- p.121Chapter 7.4 --- Computation overhead using the DC coefficient restoration scheme --- p.134Chapter 7.5 --- Summary --- p.134Chapter 8 --- Conclusions and Discussions --- p.136Chapter A --- Fundamental definitions --- p.144Chapter B --- Irreducibility by associated directed graph --- p.146Chapter B.1 --- Irreducibility and associated directed graph --- p.146Chapter B.2 --- Derivation of irreducibility --- p.147Chapter B.3 --- Multiple blocks selection --- p.149Chapter B.4 --- Irreducibility with edge selection --- p.151Chapter C --- Sample images --- p.153Bibliography --- p.15

Progressive Lossless Image Compression Using Image Decomposition and Context Quantization

Lossless image compression has many applications, for example, in medical imaging, space photograph and film industry. In this thesis, we propose an efficient lossless image compression scheme for both binary images and gray-scale images. The scheme first decomposes images into a set of progressively refined binary sequences and then uses the context-based, adaptive arithmetic coding algorithm to encode these sequences. In order to deal with the context dilution problem in arithmetic coding, we propose a Lloyd-like iterative algorithm to quantize contexts. Fixing the set of input contexts and the number of quantized contexts, our context quantization algorithm iteratively finds the optimum context mapping in the sense of minimizing the compression rate. Experimental results show that by combining image decomposition and context quantization, our scheme can achieve competitive lossless compression performance compared to the JBIG algorithm for binary images, and the CALIC algorithm for gray-scale images. In contrast to CALIC, our scheme provides the additional feature of allowing progressive transmission of gray-scale images, which is very appealing in applications such as web browsing

Proceedings of the Eighth Workshop on Information Theoretic Methods in Science and Engineering

Proceedings of the Eighth Workshop on Information Theoretic Methods in Science and Engineering (WITMSE 2015) held in Copenhagen, Denmark, 24-26 June 2015; published in the series of the Department of Computer Science, University of Helsinki.Peer reviewe

Speech Enhancement Exploiting the Source-Filter Model

Imagining everyday life without mobile telephony is nowadays hardly possible. Calls are being made in every thinkable situation and environment. Hence, the microphone will not only pick up the user’s speech but also sound from the surroundings which is likely to impede the understanding of the conversational partner. Modern speech enhancement systems are able to mitigate such effects and most users are not even aware of their existence. In this thesis the development of a modern single-channel speech enhancement approach is presented, which uses the divide and conquer principle to combat environmental noise in microphone signals. Though initially motivated by mobile telephony applications, this approach can be applied whenever speech is to be retrieved from a corrupted signal. The approach uses the so-called source-filter model to divide the problem into two subproblems which are then subsequently conquered by enhancing the source (the excitation signal) and the filter (the spectral envelope) separately. Both enhanced signals are then used to denoise the corrupted signal. The estimation of spectral envelopes has quite some history and some approaches already exist for speech enhancement. However, they typically neglect the excitation signal which leads to the inability of enhancing the fine structure properly. Both individual enhancement approaches exploit benefits of the cepstral domain which offers, e.g., advantageous mathematical properties and straightforward synthesis of excitation-like signals. We investigate traditional model-based schemes like Gaussian mixture models (GMMs), classical signal processing-based, as well as modern deep neural network (DNN)-based approaches in this thesis. The enhanced signals are not used directly to enhance the corrupted signal (e.g., to synthesize a clean speech signal) but as so-called a priori signal-to-noise ratio (SNR) estimate in a traditional statistical speech enhancement system. Such a traditional system consists of a noise power estimator, an a priori SNR estimator, and a spectral weighting rule that is usually driven by the results of the aforementioned estimators and subsequently employed to retrieve the clean speech estimate from the noisy observation. As a result the new approach obtains significantly higher noise attenuation compared to current state-of-the-art systems while maintaining a quite comparable speech component quality and speech intelligibility. In consequence, the overall quality of the enhanced speech signal turns out to be superior as compared to state-of-the-art speech ehnahcement approaches.Mobiltelefonie ist aus dem heutigen Leben nicht mehr wegzudenken. Telefonate werden in beliebigen Situationen an beliebigen Orten geführt und dabei nimmt das Mikrofon nicht nur die Sprache des Nutzers auf, sondern auch die Umgebungsgeräusche, welche das Verständnis des Gesprächspartners stark beeinflussen können. Moderne Systeme können durch Sprachverbesserungsalgorithmen solchen Effekten entgegenwirken, dabei ist vielen Nutzern nicht einmal bewusst, dass diese Algorithmen existieren. In dieser Arbeit wird die Entwicklung eines einkanaligen Sprachverbesserungssystems vorgestellt. Der Ansatz setzt auf das Teile-und-herrsche-Verfahren, um störende Umgebungsgeräusche aus Mikrofonsignalen herauszufiltern. Dieses Verfahren kann für sämtliche Fälle angewendet werden, in denen Sprache aus verrauschten Signalen extrahiert werden soll. Der Ansatz nutzt das Quelle-Filter-Modell, um das ursprüngliche Problem in zwei Unterprobleme aufzuteilen, die anschließend gelöst werden, indem die Quelle (das Anregungssignal) und das Filter (die spektrale Einhüllende) separat verbessert werden. Die verbesserten Signale werden gemeinsam genutzt, um das gestörte Mikrofonsignal zu entrauschen. Die Schätzung von spektralen Einhüllenden wurde bereits in der Vergangenheit erforscht und zum Teil auch für die Sprachverbesserung angewandt. Typischerweise wird dabei jedoch das Anregungssignal vernachlässigt, so dass die spektrale Feinstruktur des Mikrofonsignals nicht verbessert werden kann. Beide Ansätze nutzen jeweils die Eigenschaften der cepstralen Domäne, die unter anderem vorteilhafte mathematische Eigenschaften mit sich bringen, sowie die Möglichkeit, Prototypen eines Anregungssignals zu erzeugen. Wir untersuchen modellbasierte Ansätze, wie z.B. Gaußsche Mischmodelle, klassische signalverarbeitungsbasierte Lösungen und auch moderne tiefe neuronale Netzwerke in dieser Arbeit. Die so verbesserten Signale werden nicht direkt zur Sprachsignalverbesserung genutzt (z.B. Sprachsynthese), sondern als sogenannter A-priori-Signal-zu-Rauschleistungs-Schätzwert in einem traditionellen statistischen Sprachverbesserungssystem. Dieses besteht aus einem Störleistungs-Schätzer, einem A-priori-Signal-zu-Rauschleistungs-Schätzer und einer spektralen Gewichtungsregel, die üblicherweise mit Hilfe der Ergebnisse der beiden Schätzer berechnet wird. Schließlich wird eine Schätzung des sauberen Sprachsignals aus der Mikrofonaufnahme gewonnen. Der neue Ansatz bietet eine signifikant höhere Dämpfung des Störgeräuschs als der bisherige Stand der Technik. Dabei wird eine vergleichbare Qualität der Sprachkomponente und der Sprachverständlichkeit gewährleistet. Somit konnte die Gesamtqualität des verbesserten Sprachsignals gegenüber dem Stand der Technik erhöht werden

Designing new network adaptation and ATM adaptation layers for interactive multimedia applications

Multimedia services, audiovisual applications composed of a combination of discrete and continuous data streams, will be a major part of the traffic flowing in the next generation of high speed networks. The cornerstones for multimedia are Asynchronous Transfer Mode (ATM) foreseen as the technology for the future Broadband Integrated Services Digital Network (B-ISDN) and audio and video compression algorithms such as MPEG-2 that reduce applications bandwidth requirements. Powerful desktop computers available today can integrate seamlessly the network access and the applications and thus bring the new multimedia services to home and business users. Among these services, those based on multipoint capabilities are expected to play a major role.    Interactive multimedia applications unlike traditional data transfer applications have stringent simultaneous requirements in terms of loss and delay jitter due to the nature of audiovisual information. In addition, such stream-based applications deliver data at a variable rate, in particular if a constant quality is required.    ATM, is able to integrate traffic of different nature within a single network creating interactions of different types that translate into delay jitter and loss. Traditional protocol layers do not have the appropriate mechanisms to provide the required network quality of service (QoS) for such interactive variable bit rate (VBR) multimedia multipoint applications. This lack of functionalities calls for the design of protocol layers with the appropriate functions to handle the stringent requirements of multimedia.    This thesis contributes to the solution of this problem by proposing new Network Adaptation and ATM Adaptation Layers for interactive VBR multimedia multipoint services.    The foundations to build these new multimedia protocol layers are twofold; the requirements of real-time multimedia applications and the nature of compressed audiovisual data.    On this basis, we present a set of design principles we consider as mandatory for a generic Multimedia AAL capable of handling interactive VBR multimedia applications in point-to-point as well as multicast environments. These design principles are then used as a foundation to derive a first set of functions for the MAAL, namely; cell loss detection via sequence numbering, packet delineation, dummy cell insertion and cell loss correction via RSE FEC techniques.    The proposed functions, partly based on some theoretical studies, are implemented and evaluated in a simulated environment. Performances are evaluated from the network point of view using classic metrics such as cell and packet loss. We also study the behavior of the cell loss process in order to evaluate the efficiency to be expected from the proposed cell loss correction method. We also discuss the difficulties to map network QoS parameters to user QoS parameters for multimedia applications and especially for video information. In order to present a complete performance evaluation that is also meaningful to the end-user, we make use of the MPQM metric to map the obtained network performance results to a user level. We evaluate the impact that cell loss has onto video and also the improvements achieved with the MAAL.    All performance results are compared to an equivalent implementation based on AAL5, as specified by the current ITU-T and ATM Forum standards.    An AAL has to be by definition generic. But to fully exploit the functionalities of the AAL layer, it is necessary to have a protocol layer that will efficiently interface the network and the applications. This role is devoted to the Network Adaptation Layer.    The network adaptation layer (NAL) we propose, aims at efficiently interface the applications to the underlying network to achieve a reliable but low overhead transmission of video streams. Since this requires an a priori knowledge of the information structure to be transmitted, we propose the NAL to be codec specific.    The NAL targets interactive multimedia applications. These applications share a set of common requirements independent of the encoding scheme used. This calls for the definition of a set of design principles that should be shared by any NAL even if the implementation of the functions themselves is codec specific. On the basis of the design principles, we derive the common functions that NALs have to perform which are mainly two; the segmentation and reassembly of data packets and the selective data protection.    On this basis, we develop an MPEG-2 specific NAL. It provides a perceptual syntactic information protection, the PSIP, which results in an intelligent and minimum overhead protection of video information. The PSIP takes advantage of the hierarchical organization of the compressed video data, common to the majority of the compression algorithms, to perform a selective data protection based on the perceptual relevance of the syntactic information.    The transmission over the combined NAL-MAAL layers shows significant improvement in terms of CLR and perceptual quality compared to equivalent transmissions over AAL5 with the same overhead.    The usage of the MPQM as a performance metric, which is one of the main contributions of this thesis, leads to a very interesting observation. The experimental results show that for unexpectedly high CLRs, the average perceptual quality remains close to the original value. The economical potential of such an observation is very important. Given that the data flows are VBR, it is possible to improve network utilization by means of statistical multiplexing. It is therefore possible to reduce the cost per communication by increasing the number of connections with a minimal loss in quality.    This conclusion could not have been derived without the combined usage of perceptual and network QoS metrics, which have been able to unveil the economic potential of perceptually protected streams.    The proposed concepts are finally tested in a real environment where a proof-of-concept implementation of the MAAL has shown a behavior close to the simulated results therefore validating the proposed multimedia protocol layers