In the last decades, air traffic spread more and more in the world, connecting more and more places. At the same time, the need to manage all the flights correctly and securely increased. Air traffic authorities imposed and updated several standards for the air traffic management (ATM) system, keeping in pace with the growing traffic flow. To achieve this, special voice communication systems (VCS) were developed. They ensure the communication between the pilots and the operators from the ground control centers. When a communication is initiated between the aircraft’s pilot and the ground air traffic control operator, various systems are used. The pilot speaks through the aircraft’s radio station and the signal is received by several ground radio stations. Then, the signal from each ground radio station arrives on different paths to the control center. Here one of the received ...

In this work, a novel set of robust watermark-based error concealment (WEC) algorithms are proposed. Watermarking is used to introduce redundancy to the transmitted data with little or no increase in its bit rate during transmission. The proposed algorithms involve generating a low resolution version of a video frame and seamlessly embedding it as a watermark in the frame itself during encoding. At the receiver, the watermark is extracted from the reconstructed frame and the lost information is recovered using the extracted watermark signal, thus enhancing its perceptual quality. Three DCT-based spread spectrum watermark embedding techniques are presented in this work. The first technique uses a multiplicative Gaussian pseudo-noise with a pre-defined spreading gain and fixed chip rate. The second one is its adaptively scaled version and the third technique uses informed watermarking. Two versions of the low resolution reference, ...

Single-channel speech enhancement refers to the reduction of noise signal components in a single-channel signal composed of both speech and noise. Spectral speech enhancement methods are among the most popular approaches to solving this problem. Since the short-time spectral amplitude has been identified as a highly perceptually relevant quantity, most conventional approaches rely on processing the amplitude spectrum only, ignoring any information that may be contained in the spectral phase. As a consequence, the noisy short-time spectral phase is neither enhanced for the purpose of signal reconstruction nor is it used for refining short-time spectral amplitude estimates. This thesis investigates the use of the spectral phase and its structure in algorithms for single-channel speech enhancement. This includes the analysis of the spectral phase in the context of theoretically optimal speech estimators. The resulting knowledge is exploited in formulating single-channel speech ...

The widespread use of mobile internet and smart applications has led to an explosive growth in mobile data traffic. With the rise of smart homes, smart buildings, and smart cities, this demand is ever growing since future communication systems will require the integration of multiple networks serving diverse sectors, domains and applications, such as multimedia, virtual or augmented reality, machine-to-machine (M2M) communication / the Internet of things (IoT), automotive applications, and many more. Therefore, in the future, the communication systems will not only be required to provide Gbps wireless connectivity but also fulfill other requirements such as low latency and massive machine type connectivity while ensuring the quality of service. Without significant technological advances to increase the system capacity, the existing telecommunications infrastructure will be unable to support these multi-dimensional requirements. This poses an important demand for suitable waveforms with ...

The envisioned rapid and exponential increase of wireless data traffic demand in the next years imposes rethinking current wireless cellular networks due to the scarcity of the available spectrum. In this regard, three main drivers are considered to increase the capacity of today's most advanced (4G systems) and future (5G systems and beyond) cellular networks: i) use more bandwidth (more Hz) through spectral aggregation, ii) enhance the spectral efficiency per base station (BS) (more bits/s/Hz/BS) by using multiple antennas at BSs and users (i.e. MIMO systems), and iii) increase the density of BSs (more BSs/km2) through a dense and heterogeneous deployment (known as dense heterogeneous cellular networks). We focus on the last two drivers. First, the use of multi-antenna systems allows exploiting the spatial dimension for several purposes: improving the capacity of a conventional point-to-point wireless link, increasing the number ...

In this thesis, a number of novel watermarking techniques for different 3D representations are presented. A novel watermarking method is proposed for the mono-view video, which might be interpreted as the basic implicit representation of 3D scenes. The proposed method solves the common flickering problem in the existing video watermarking schemes by means of adjusting the watermark strength with respect to temporal contrast thresholds of human visual system (HVS), which define the maximum invisible distortions in the temporal direction. The experimental results indicate that the proposed method gives better results in both objective and subjective measures, compared to some recognized methods in the literature. The watermarking techniques for the geometry and image based representations of 3D scenes, denoted as 3D watermarking, are examined and classified into three groups, as 3D-3D, 3D-2D and 2D-2D watermarking, in which the pair of symbols ...

The growing risk of privacy violation and espionage associated with the rapid spread of mobile communications renewed interest in the original concept of sending encrypted voice as audio signal over arbitrary voice channels. The usual methods used for encrypted data transmission over analog telephony turned out to be inadequate for modern vocal links (cellular networks, VoIP) equipped with voice compression, voice activity detection, and adaptive noise suppression algorithms. The limited available bandwidth, nonlinear channel distortion, and signal fadings motivate the investigation of a dedicated, joint approach for speech encoding and encryption adapted to modern noisy voice channels. This thesis aims to develop, analyze, and validate secure and efficient schemes for real-time speech encryption and transmission via modern voice channels. In addition to speech encryption, this study covers the security and operational aspects of the whole voice communication system, as this ...

Steganography is the science that involves communicating secret data in an appropriate multimedia carrier, e.g., image, audio and video files. It comes under the assumption that if the feature is visible, the point of attack is evident, thus the goal here is always to conceal the very existence of the embedded data. It does not replace cryptography but rather boosts the security using its obscurity features. Steganography has various useful applications. However, like any other science it can be used for ill intentions. It has been propelled to the forefront of current security techniques by the remarkable growth in computational power, the increase in security awareness, e.g., individuals, groups, agencies, government and through intellectual pursuit. Steganography’s ultimate objectives, which are undetectability, robustness, resistance to various image processing methods and compression, and capacity of the hidden data, are the main factors ...

In the early days of speech synthesis research, understanding voice production has attracted the attention of scientists with the goal of producing intelligible speech. Later, the need to produce more natural voices led researchers to use prerecorded voice databases, containing speech units, reassembled by a concatenation algorithm. With the outgrowth of computer capacities, the length of units increased, going from diphones to non-uniform units, in the so-called unit selection framework, using a strategy referred to as 'take the best, modify the least'. Today the new challenge in voice synthesis is the production of expressive speech or singing. The mainstream solution to this problem is based on the “there is no data like more data” paradigm: emotionspecific databases are recorded and emotion-specific units are segmented. In this thesis, we propose to restart the expressive speech synthesis problem, from its original voice ...

The main focus of this thesis is the problem of geometric distortion in image and video watermarking. In this thesis we discuss the two aspects of the geometric distortion problem, namely the watermark desynchronization aspect and the perceptual quality assessment aspect. Furthermore, this thesis also discusses the challenges of watermarking data compressed in low bit-rates. The main contributions of this thesis are: A watermarking algorithm suitable for low bit-rate video has been proposed. Two different approaches has been proposed to deal with the watermark desynchronization problem. A novel approach has been proposed to quantify the perceptual quality impact of geometric distortion.

Humans without larynx need to use a substitution voice to re-obtain speech. The electro-larynx (EL) is a widely used device but is known for its unnatural and monotonic speech quality. Previous research tackled these problems, but until now no significant improvements could be reported. The EL speech system is a complex system including hardware (artificial excitation source or sound transducer) and software (control and generation of the artificial excitation signal). It is not enough to consider one separated problem, but all aspects of the EL speech system need to be taken into account. In this thesis we would like to push forward the boundaries of the conventional EL device towards a new bionic electro-larynx speech system. We formulate two overall scenarios: a closed-loop scenario, where EL speech is excited and simultaneously recorded using an EL speech system, and the artificial ...

In scalable image/video coding, high resolution content is encoded to the highest visual quality and the bit-streams are adapted to cater various communication channels, display devices and usage requirements. These content adaptations, which include quality, resolution and frame rate scaling may also affect the content protection data, such as, watermarks and are considered as a potential watermark attack. In this thesis, research on robust watermarking techniques for scalable coded image and video, are proposed and the improvements in robustness against various content adaptation attacks, such as, JPEG 2000 for image and Motion JPEG 2000, MC-EZBC and H.264/SVC for video, are reported. The spread spectrum domain, particularly wavelet-based image watermarking schemes often provides better robustness to compression attacks due to its multi-resolution decomposition and hence chosen for this work. A comprehensive and comparative analysis of the available wavelet-based watermarking schemes,is performed ...

The appetite amongst consumers for ever higher data-rates seems insatiable. This booming market presents a huge opportunity for telephone and cable operators. It also presents a challenge: the delivery of broadband services to millions of customers across sparsely populated areas. Fully bre-based networks, whilst technically the most advanced solution, are prohibitively expensive to deploy. Digital subscriber lines (DSL) provide an alternative solution. Seen as a stepping-stone to a fully bre-based network, DSL operates over telephone lines that are already in place, minimizing the cost of deployment. The basic principle behind DSL technology is to increase data-rate by widening the transmission bandwidth. Unfortunately, operating at high frequencies, in a medium originally designed for voice-band transmission, leads to crosstalk between the di erent DSLs. Crosstalk is typically 10-15 dB larger than the background noise and is the dominant source of performance degradation ...

In many speech applications such as hands-free telephony or voice-controlled home assistants, the distance between the user and the recording microphones can be relatively large. In such a far-field scenario, the recorded microphone signals are typically corrupted by noise and reverberation, which may severely degrade the performance of speech recognition systems and reduce intelligibility and quality of speech in communication applications. In order to limit these effects, speech enhancement algorithms are typically applied. The main objective of this thesis is to develop novel speech enhancement algorithms for noisy and reverberant environments and signal-based measures to evaluate these algorithms, focusing on solutions that are applicable in realistic scenarios. First, we propose a single-channel speech enhancement algorithm for joint noise and reverberation reduction. The proposed algorithm uses a spectral gain to enhance the input signal, where the gain is computed using a ...

Despite the outstanding progress made on automatic speech recognition (ASR) throughout the last decades, noise-robust ASR still poses a challenge. Tackling with acoustic noise in ASR systems is more important than ever before for a twofold reason: 1) ASR technology has begun to be extensively integrated in intelligent mobile devices (IMDs) such as smartphones to easily accomplish different tasks (e.g. search-by-voice), and 2) IMDs can be used anywhere at any time, that is, under many different acoustic (noisy) conditions. On the other hand, with the aim of enhancing noisy speech, IMDs have begun to embed small microphone arrays, i.e. microphone arrays comprised of a few sensors close each other. These multi-sensor IMDs often embed one microphone (usually at their rear) intended to capture the acoustic environment more than the speaker’s voice. This is the so-called secondary microphone. While classical microphone ...