Abstract Nowadays, content is increasingly in digital form and distributed in the Internet. The ease of making perfect copies of the digital content has created a need to develop a means to protect it. Digital rights management (DRM) relates to systems designed to protect the intellectual property rights of the digital content. The DRM systems try to enable a secure distribution of digital content to the users and to prevent the unauthorized copying, usage, and distribution of the content. This is usually done in practice using encryption and digital watermarking techniques. This thesis concentrates on the problem of protecting and distributing multimedia content securely in mobile environment. The research objectives are: (1) to design an overall DRM architecture which allows an easy content distribution to the user in mobile environment; (2) to develop protection methods that can be used in mobile devices with limited computational capabilities to prevent unauthorized usage of the audio content; (3) to create methods for managing and enforcing the user’s rights and restrictions to the content usage; (4) to study a method for providing the users with an easy access to new digital content and services. The research is carried out by first developing an overall DRM platform to mobile environment. The experimental prototype of the platform is implemented on server side to PC environment and the client runs on a mobile phone. The platform is used to test the functionality and complexity of the content protection methods developed which are based on digital watermarking and encryption techniques. The main results of the thesis are: (1) a DRM platform for mobile devices that supports peer-to-peer networking and license negotiation; (2) audio protection methods utilizing digital watermarking and encryption techniques which support content superdistribution and content preview; (3) methods for counting offline how many times content has been played on the user’s terminal using watermarking and hash chains; (4) a method for adding metadata, such as a web link, into audio content, so that it survives digital to analog to digital transformation and recording with a mobile phone.

Abstract Broadband communication networks and multimedia data available in a digital format opened many challenges and opportunities for innovation. Versatile and simple-to-use software and decreasing prices of digital devices have made it possible for consumers from all around the world to create and exchange multimedia data. Broadband Internet connections and near error-free transmission of data facilitate people to distribute large multimedia files and make identical digital copies of them. A perfect reproduction in digital domain have promoted the protection of intellectual ownership and the prevention of unauthorized tampering of multimedia data to become an important technological and research issue. Digital watermarking has been proposed as a new, alternative method to enforce intellectual property rights and protect digital media from tampering. Digital watermarking is defined as imperceptible, robust and secure communication of data related to the host signal, which includes embedding into and extraction from the host signal. The main challenge in digital audio watermarking and steganography is that if the perceptual transparency parameter is fixed, the design of a watermark system cannot obtain high robustness and a high watermark data rate at the same time. In this thesis, we address three research problems on audio watermarking: First, what is the highest watermark bit rate obtainable, under the perceptual transparency constraint, and how to approach the limit? Second, how can the detection performance of a watermarking system be improved using algorithms based on communications models for that system? Third, how can overall robustness to attacks to a watermark system be increased using attack characterization at the embedding side? An approach that combined theoretical consideration and experimental validation, including digital signal processing, psychoacoustic modeling and communications theory, is used in developing algorithms for audio watermarking and steganography. The main results of this study are the development of novel audio watermarking algorithms, with the state-of-the-art performance and an acceptable increase in computational complexity. The algorithms' performance is validated in the presence of the standard watermarking attacks. The main technical solutions include algorithms for embedding high data rate watermarks into the host audio signal, using channel models derived from communications theory for watermark transmission and the detection and modeling of attacks using attack characterization procedure. The thesis also includes a thorough review of the state-of-the-art literature in the digital audio watermarking.

Due to the accelerated growth in digital music distribution, it becomes easy to modify, intercept, and distribute material illegally. To overcome the urgent need for copyright protection against piracy, several audio watermarking schemes have been proposed and implemented. These digital audio watermarking schemes have the purpose of embedding inaudible information within the host file to cover copyright and authentication issues. This thesis proposes an audio watermarking model using MATLAB® and Simulink® software for 1K and 2K fast Fourier transform (FFT) lengths. The watermark insertion process is performed in the frequency domain to guarantee the imperceptibility of the watermark to the human auditory system. Additionally, the proposed audio watermarking model was implemented in a Cyclone® II FPGA device from Altera® using the Altera® DSP Builder tool and MATLAB/Simulink® software. To evaluate the performance of the proposed audio watermarking scheme, effectiveness and fidelity performance tests were conducted for the proposed software and hardware-in-the-loop based audio watermarking model.

In spreads pectrum (SS)watermarking of audio signals, since the watermark acts as an additive noise to the host audio signal, the most important challenge is to maintain perceptual transparency. Human perception is a very sensitive apparatus, yet can be exploited to hide some information, reliably. SS watermark embedding has been proposed, in which psycho-acoustically shaped pseudo-random sequences are embedded directly into the time domain audio signal. However, these watermarking schemes use informed detection, in which the original signal is assumed available to the watermark detector. Blind detection of psycho-acoustically shaped SS watermarking is not well addressed in the literature. The problem is still interesting, because, blind detection is more practical for audio signals and, psycho-acoustically shaped watermarks embedding offers the maximum possible watermark energy under requirements of perceptual transparency. In this thesis we study the blind detection of psycho-acoustically shaped SS watermarks in time domain audio signals. We focus on a class of watermark sequences known as random phase watermarks, where the watermark magnitude spectrum is defined by the perceptual criteria and the randomness of the sequence lies in their phase spectrum. Blind watermark detectors, which do not have access to the original host signal, may seem handicapped, because an approximate watermark has to be re-derived from the watermarked signal. Since the comparison of blind detection with fully informed detection is unfair, a hypothetical detection scheme, denoted as semi-blind detection, is used as a reference benchmark. In semi-blind detection, the host signal as such is not available for detection, but it is assumed that sufficient information is available for deriving the exact watermark, which could be embedded in the given signal. Some reduction in performance is anticipated in blind detection over the semi-blind detection. Our experiments revealed that the statistical performance of the blind detector is better than that of the semi-blind detector. We analyze the watermark-to-host correlation (WHC) of random phase watermarks, and the results indicate that WHC is higher when a legitimate watermark is present in the audio signal, which leads to better detection performance. Based on these findings, we attempt to harness this increased correlation in order to further improve the performance. The analysis shows that uniformly distributed phase difference (between the host signal and the watermark) provides maximum advantage. This property is verified through experimentation over a variety of audio signals. In the second part, the correlated nature of audio signals is identified as a potential threat to reliable blind watermark detection, and audio pre-whitening methods are suggested as a possible remedy. A direct deterministic whitening (DDW) scheme is derived, from the frequency domain analysis of the time domain correlation process. Our experimental studies reveal that, the Savitzky-Golay Whitening (SGW), which is otherwise inferior to DDW technique, performs better when the audio signal is predominantly low pass. The novelty of this work lies in exploiting the complementary nature of the two whitening techniques and combining them to obtain a hybrid whitening (HbW) scheme. In the hybrid scheme the DDW and SGW techniques are selectively applied, based on short time spectral characteristics of the audio signal. The hybrid scheme extends the reliability of watermark detection to a wider range of audio signals. We also discuss enhancements to the HbW technique for robustness to temporal offsets and filtering. Robustness of SS watermark blind detection, with hybrid whitening, is determined through a set of experiments and the results are presented. It is seen that the watermarking scheme is robust to common signal processing operations such as additive noise, filtering, lossy compression, etc.

This Thesis deals with a method to enforce the intellectual property rights and protect digital media from tampering – Digital Audio Watermarking. The main aim of this work is implement an audio watermarking algorithm. The theoretical part defined basic terms, methods and processes, which are used in this area. The practical part shows a process of embedding the digital signature into a host signal and her backward extraction. The embedding rule used spread spectrum technique and a psychoacoustic model. The implemented psychoacoustic model involves two properties of the human auditory system which are frequency masking and representation the frequency scale on limited bands called critical bands. The model is relatively new and based on the DWPT. In terms of above model is then the digital watermark embedded in the wavelet domain. This algorithm is implemented in technical software MATLAB. One part of this work focuses on robustness tests of the algorithm. Common signal processing modifications are applied to the watermarked audio as follows: Cutting of the audio, re-sampling, lossy compression, filtering, equalization, modulation effects, noise addition. The last part of the thesis presents subjective and objective methods usable in order to judge the influence of watermarking embedding on the quality of audio tracks called transparency.

In this work is presented technique for embedding digital watermark in digital audio signals. Digital watermark must be imperceptible and should be robust against attacks and other types of distortion. Algorithm is implemented for embedding digital watermark using technique spread-spectrum and psychoacoustic model ISO-MPEG I layer I. Robustness was tested for filtering signal, MP3 compression and resample method.

L'objectif de cette thèse est d'étudier l'idée du tatouage dans le traitement du son.Les recherches en tatouage audio se sont principalement tournées vers des applications sécuritaires ou de transmission de données auxiliaires. Une des applications visées par ce concept consiste à améliorer la qualité du signal hôte ayant subi des transformations et ceci en exploitant l'information qu'il véhicule. Le tatouage audio est donc considéré comme mémoire porteuse d'informations sur le signal originel. La compression à bas débit des signaux audio est une des applications visée par ce concept. Dans ce cadre, deux objectifs sont proposés : la réduction du pré-écho et de l'amollissement d'attaque, deux phénomènes introduits par les codeurs audio perceptifs, en particulier les codeurs AAC et MP3; la préservation de l'harmonicité des signaux audio dégradée par les codeurs perceptifs à extension de bande, en particulier le codeur HE-AAC.La première partie de ce manuscrit présente les principes de base des systèmes de codage bas débit et étudie les différentes distorsions introduites par ces derniers. Fondées sur cette étude, deux solutions sont proposées. La première, visant principalement la réduction du pré-écho, consiste à corriger l'enveloppe temporelle du signal après réception en exploitant la connaissance a priori de l'enveloppe temporelle du signal original, supposée transmise par un canal auxiliaire à faible débit (< 500 bits/s). La seconde solution vise à corriger les ruptures d'harmonicité générées par les codeurs à extension de bande. Ce phénomène touche essentiellement les signaux fortement harmoniques (exemple : violon) et est perçu comme une dissonance. Une préservation de l'harmonicité des signaux audio par des opérations de translation spectrale est alors proposée, les paramètres étant là encore transmis par un canal auxiliaire à faible débit.La seconde partie de ce document est consacrée à l'intégration du tatouage audio dans les techniques de renforcement de la qualité des signaux audio précitées. Dans ce contexte, le tatouage audio remplace le canal auxiliaire précédent et œuvre comme une mémoire du signal originel, porteuse d'informations nécessaires pour la correction d'harmonicité et la réduction de pré-écho. Cette seconde partie a été précédée par une étape approfondie de l'évaluation des performances de la technique de tatouage adoptée en terme de robustesse à la compression MPEG (MP3, AAC et aacPlus). / The goal of this thesis is to explore the idea of watermark for sound enhancement. Classically, watermark schemes are oriented towards security applications or maximization of the transmitted bit rates. Our approach is completely different. Our goal is to study how an audio watermarking can improve the quality of the host audio signal by exploiting the information it conveys. The audio watermarking is considered as a memory that carries information about the original signal.The low bitrate compression of audio signals is one of the applications covered by this concept. In this context, two objectives are proposed: reducing the pre-echo and the attack softening, two phenomena introduced by the perceptual audio coders, particularly AAC and MP3 encoders ; preserving the harmonicity of audio signals, distorted by coders with bandwidth extension, especially HE-AAC encoder. These coders are limited in the reconstruction of the high-frequency spectrum mainly because of the potential unpredictability of the fine structure of the latter, as well as imperfect indicators of tonal to noise.The first part of this manuscript presents the basic principles of low rate coding systems and studies the various distortions introduced by the latter. Based on this study, two solutions are proposed. The first one, principally aimed at reducing the pre-echo, consist in correcting the time envelope of the signal after reception by exploiting the prior knowledge of the temporal envelope of the original signal, which is assumed transmitted by an auxiliary channel at low bitrates (<500 bps). The second solution is to correct the harmonicity generated by coders with bandwidth extension. This primarily affects strongly harmonic signals (e.g. violin) and is perceived as a dissonance. We propose then to preserve the harmonicity of audio signals by spectral translations. The parameters being passed again by an auxiliary channel at low bitrates.The second part of this document is dedicated to the integration of audio watermarking techniques in the solution presented in the first part. In this context, the audio watermarking replaces the previous auxiliary channel and is regarded as a memory of the original signal, carrying information necessary for the correction of harmonicity and the pre-echo reduction.

MPEG audio compression techniques significantly reduces the storage and transmission requirements for high quality digital audio. However, compression complicates the processing of audio in many applications. If a compressed audio signal is to be processed, a direct method would be to decode the compressed signal, process the decoded signal and re-encode it. This is computationally expensive due to the complexity of the MPEG filter bank. This thesis deals with processing of MPEG compressed audio. The main contributions of this thesis are a) Extracting wavelet coefficients in the MPEG compressed domain. b) Wavelet based pitch extraction in MPEG compressed domain. c) Time Scale Modifications of MPEG audio. d) Watermarking of MPEG audio. The research contributions starts with a technique for calculating several levels of wavelet coefficients from the output of the MPEG analysis filter bank. The technique exploits the toeplitz structure which arises when the MPEG and wavelet filter banks are represented in a matrix form, The computational complexity for extracting several levels of wavelet coefficients after decoding the compressed signal and directly from the output of the MPEG analysis filter bank are compared. The proposed technique is found to be computationally efficient for extracting higher levels of wavelet coefficients. Extracting pitch in the compressed domain becomes essential when large multimedia databases need to be indexed. For example one may be interested in listening to a particular speaker or to listen to male female audio segments in a multimedia document. For this application, pitch information is one of the very basic and important features required. Pitch is basically the time interval between two successive glottal closures. Glottal closures are accompanied by sharp transients in the speech signal which in turn gives rise to a local maxima in the wavelet coefficients. Pitch can be calculated by finding the time interval between two successive maxima in the wavelet coefficients. It is shown that the computational complexity for extracting pitch in the compressed domain is less than 7% of the uncompressed domain processing. An algorithm for extracting pitch in the compressed domain is proposed. The result of this algorithm for synthetic signals, and utterances of words by male/female is reported. In a number of important applications, one needs to modify an audio signal to render it more useful than its original. Typical applications include changing the time evolution of an audio signal (increase or decrease the rate of articulation of a speaker),or to adapt a given audio sequence to a given video sequence. In this thesis, time scale modifications are obtained in the subband domain such that when the modified subband signals are given to the MPEG synthesis filter bank, the desired time scale modification of the decoded signal is achieved. This is done by making use of sinusoidal modeling [I]. Here, each of the subband signal is modeled in terms of parameters such as amplitude phase and frequencies and are subsequently synthesised by using these parameters with Ls = k La where Ls is the length of the synthesis window , k is the time scale factor and La is the length of the analysis window. As the PCM version of the time scaled signal is not available, psychoacoustic model based bit allocation cannot be used. Hence a new bit allocation is done by using a subband coding algorithm. This method has been satisfactorily tested for time scale expansion and compression of speech and music signals. The recent growth of multimedia systems has increased the need for protecting digital media. Digital watermarking has been proposed as a method for protecting digital documents. The watermark needs to be added to the signal in such a way that it does not cause audible distortions. However the idea behind the lossy MPEC encoders is to remove or make insignificant those portions of the signal which does not affect human hearing. This renders the watermark insignificant and hence proving ownership of the signal becomes difficult when an audio signal is compressed. The existing compressed domain methods merely change the bits or the scale factors according to a key. Though simple, these methods are not robust to attacks. Further these methods require original signal to be available in the verification process. In this thesis we propose a watermarking method based on spread spectrum technique which does not require original signal during the verification process. It is also shown to be more robust than the existing methods. In our method the watermark is spread across many subband samples. Here two factors need to be considered, a) the watermark is to be embedded only in those subbands which will make the addition of the noise inaudible. b) The watermark should be added to those subbands which has sufficient bit allocation so that the watermark does not become insignificant due to lack of bit allocation. Embedding the watermark in the lower subbands would cause distortion and in the higher subbands would prove futile as the bit allocation in these subbands are practically zero. Considering a11 these factors, one can introduce noise to samples across many frames corresponding to subbands 4 to 8. In the verification process, it is sufficient to have the key/code and the possibly attacked signal. This method has been satisfactorily tested for robustness to scalefactor, LSB change and MPEG decoding and re-encoding.