This thesis deals with lossless and losing methods of data compressions and their possible applications in the measurement engineering. In the first part of the thesis there is a theoretical elaboration which informs the reader about the basic terminology, the reasons of data compression, the usage of data compression in standard practice and the division of compression algorithms. The practical part of thesis deals with the realization of the compress algorithms in Matlab and LabWindows/CVI.

This thesis is focused on the compression algorithm's analysis of MJPEG format and its implementation in FPGA chip. Three additional video bitstream reduction methods have been evaluated for real-time low latency applications of MJPEG format. These methods are noise filtering, inter-frame encoding and lowering video's quality. Based on this analysis, a MJPEG codec has been designed for implementation into FPGA chip XC6SLX45, from Spartan-6 family.

Modified discrete cosine transform (MDCT) derived from DCT IV has emerged as the most suitable choice for transform domain audio coding applications due to its time domain alias cancellation property and de-correlation capability. In the present research work, we focus on MDCT domain analysis of audio signals for compression and other applications. We have derived algorithms for linear filtering in DCT IV and DST IV domains for symmetric and non-symmetric filter impulse responses. These results are also extended to MDCT and MDST domains which have the special property of time domain alias cancellation. We also derive filtering algorithms for the DCT II and DCT III domains. Comparison with other methods in the literature shows that, the new algorithm developed is computationally MAC efficient. These results are useful for MDCT domain audio processing such as reverb synthesis, without having to reconstruct the time domain signal and then perform the necessary filtering operations. In audio coding, the psychoacoustic model plays a crucial role and is used to estimate the masking thresholds for adaptive bit-allocation. Transparent quality audio coding is possible if the quantization noise is kept below the masking threshold for each frame. In the existing methods, the masking threshold is calculated using the DFT of the signal frame separately for MDCT domain adaptive quantization. We have extended the spectral integration based psychoacoustic model proposed for sinusoidal modeling of audio signals to the MDCT domain. This has been possible because of the detailed analysis of the relation between DFT and MDCT; we interpret the MDCT coefficients as co-sinusoids and then apply the sinusoidal masking model. The validity of the masking threshold so derived is verified through listening tests as well as objective measures. Parametric coding techniques are used for low bit rate encoding of multi-channel audio such as 5.1 format surround audio. In these techniques, the surround channels are synthesized at the receiver using the analysis parameters of the parametric model. We develop algorithms for MDCT domain analysis and synthesis of reverberation. Integrating these ideas, a parametric audio coder is developed in the MDCT domain. For the parameter estimation, we use a novel analysis by synthesis scheme in the MDCT domain which results in better modeling of the spatial audio. The resulting parametric stereo coder is able to synthesize acceptable quality stereo audio from the mono audio channel and a side information of approximately 11 kbps. Further, an experimental audio coder is developed in the MDCT domain incorporating the new psychoacoustic model and the parametric model.

Extracting speaker-specific information from speech is of great interest to both researchers and developers alike, since speaker recognition technology finds application in a wide range of areas, primary among them being forensics and biometric security systems. Several models and techniques have been employed to extract speaker information from the speech signal. Speech production is generally modeled as an excitation source followed by a filter. Physiologically, the source corresponds to the vocal fold vibrations and the filter corresponds to the spectrum-shaping vocal tract. Vocal tract-based features like the melfrequency cepstral coefficients (MFCCs) and linear prediction cepstral coefficients have been shown to contain speaker information. However, high speed videos of the larynx show that the vocal folds of different individuals vibrate differently. Voice source (VS)-based features have also been shown to perform well in speaker recognition tasks, thereby revealing that the VS does contain speaker information. Moreover, a combination of the vocal tract and VS-based features has been shown to give an improved performance, showing that the latter contains supplementary speaker information. In this study, the focus is on extracting speaker information from the VS. The existing techniques for the same are reviewed, and it is observed that the features which are obtained by fitting a time-domain model on the VS perform poorly than those obtained by simple transformations of the VS. Here, an attempt is made to propose an alternate way of characterizing the VS to extract speaker information, and to study the merits and shortcomings of the proposed speaker-specific features. The VS cannot be measured directly. Thus, to characterize the VS, we first need an estimate of the VS, and the integrated linear prediction residual (ILPR) extracted from the speech signal is used as the VS estimate in this study. The voice source linear prediction model, which was proposed in an earlier study to obtain the ILPR, is used in this work. It is hypothesized here that a speaker’s voice may be characterized by the relative proportions of the harmonics present in the VS. The pitch synchronous discrete cosine transform (DCT) is shown to capture these, and the gross shape of the ILPR in a few coefficients. The ILPR and hence its DCT coefficients are visually observed to distinguish between speakers. However, it is also observed that they do have intra-speaker variability, and thus it is hypothesized that the distribution of the DCT coefficients may capture speaker information, and this distribution is modeled by a Gaussian mixture model (GMM). The DCT coefficients of the ILPR (termed the DCTILPR) are directly used as a feature vector in speaker identification (SID) tasks. Issues related to the GMM, like the type of covariance matrix, are studied, and it is found that diagonal covariance matrices perform better than full covariance matrices. Thus, mixtures of Gaussians having diagonal covariances are used as speaker models, and by conducting SID experiments on three standard databases, it is found that the proposed DCTILPR features fare comparably with the existing VS-based features. It is also found that the gross shape of the VS contains most of the speaker information, and the very fine structure of the VS does not help in distinguishing speakers, and instead leads to more confusion between speakers. The major drawbacks of the DCTILPR are the session and handset variability, but they are also present in existing state-of-the-art speaker-specific VS-based features and the MFCCs, and hence seem to be common problems. There are techniques to compensate these variabilities, which need to be used when the systems using these features are deployed in an actual application. The DCTILPR is found to improve the SID accuracy of a system trained with MFCC features by 12%, indicating that the DCTILPR features capture speaker information which is missed by the MFCCs. It is also found that a combination of MFCC and DCTILPR features on a speaker verification task gives significant performance improvement in the case of short test utterances. Thus, on the whole, this study proposes an alternate way of extracting speaker information from the VS, and adds to the evidence for speaker information present in the VS.

Thesis deals with multimedia signal comparison of compression options focused on video and advanced codecs. Specifically it describes the encoding and decoding of video recordings according to the MPEG standard. The theoretical part of the thesis describes characteristic properties of the video signal and justification for the need to use recording and transmission compression. There are also described methods for elimination of encoded video signal redundancy and irrelevance. Further on are discussed ways of measuring the video signal quality. A separate chapter is focused on the characteristics of currently used and promising codecs. In the practical part of the thesis were created functions in Matlab environment. These functions were implemented into graphic user interface that simulates the activity of functional blocks of the encoder and decoder. Based on user-specified input parameters it performs encoding and decoding of any given picture, composed of images in RGB format, and displays the outputs of individual functional blocks. There are implemented algorithms for the initial processing of the input sequence including sub-sampling, as well as DCT, quantization, motion compensation and their inverse operations. Separate chapters are dedicated to the realisation of codec description in the Matlab environment and to the individual processing steps output. Further on are mentioned compress algorithm comparisons and the impact of parameter change onto the final signal. The findings are summarized in conclusion.

In this work, the fast evaluation and reconstruction of multivariate trigonometric polynomials with frequencies supported on arbitrary index sets of finite cardinality is considered, where rank-1 lattices are used as spatial discretizations. The approximation of multivariate smooth periodic functions by trigonometric polynomials is studied, based on a one-dimensional FFT applied to function samples. The smoothness of the functions is characterized via the decay of their Fourier coefficients, and various estimates for sampling errors are shown, complemented by numerical tests for up to 25 dimensions. In addition, the special case of perturbed rank-1 lattice nodes is considered, and a fast Taylor expansion based approximation method is developed. One main contribution is the transfer of the methods to the non-periodic case. Multivariate algebraic polynomials in Chebyshev form are used as ansatz functions and rank-1 Chebyshev lattices as spatial discretizations. This strategy allows for using fast algorithms based on a one-dimensional DCT. The smoothness of a function can be characterized via the decay of its Chebyshev coefficients. From this point of view, estimates for sampling errors are shown as well as numerical tests for up to 25 dimensions. A further main contribution is the development of a high-dimensional sparse FFT method based on rank-1 lattice sampling, which allows for determining unknown frequency locations belonging to the approximately largest Fourier or Chebyshev coefficients of a function. / In dieser Arbeit wird die schnelle Auswertung und Rekonstruktion multivariater trigonometrischer Polynome mit Frequenzen aus beliebigen Indexmengen endlicher Kardinalität betrachtet, wobei Rang-1-Gitter (rank-1 lattices) als Diskretisierung im Ortsbereich verwendet werden. Die Approximation multivariater glatter periodischer Funktionen durch trigonometrische Polynome wird untersucht, wobei Approximanten mittels einer eindimensionalen FFT (schnellen Fourier-Transformation) angewandt auf Funktionswerte ermittelt werden. Die Glattheit von Funktionen wird durch den Abfall ihrer Fourier-Koeffizienten charakterisiert und mehrere Abschätzungen für den Abtastfehler werden gezeigt, ergänzt durch numerische Tests für bis zu 25 Raumdimensionen. Zusätzlich wird der Spezialfall gestörter Rang-1-Gitter-Knoten betrachtet, und es wird eine schnelle Approximationsmethode basierend auf Taylorentwicklung vorgestellt. Ein wichtiger Beitrag dieser Arbeit ist die Übertragung der Methoden vom periodischen auf den nicht-periodischen Fall. Multivariate algebraische Polynome in Chebyshev-Form werden als Ansatzfunktionen verwendet und sogenannte Rang-1-Chebyshev-Gitter als Diskretisierungen im Ortsbereich. Diese Strategie ermöglicht die Verwendung schneller Algorithmen basierend auf einer eindimensionalen DCT (diskreten Kosinustransformation). Die Glattheit von Funktionen kann durch den Abfall ihrer Chebyshev-Koeffizienten charakterisiert werden. Unter diesem Gesichtspunkt werden Abschätzungen für Abtastfehler gezeigt sowie numerische Tests für bis zu 25 Raumdimensionen. Ein weiterer wichtiger Beitrag ist die Entwicklung einer Methode zur Berechnung einer hochdimensionalen dünnbesetzten FFT basierend auf Abtastwerten an Rang-1-Gittern, wobei diese Methode die Bestimmung unbekannter Frequenzen ermöglicht, welche zu den näherungsweise größten Fourier- oder Chebyshev-Koeffizienten einer Funktion gehören.

In this work, the fast evaluation and reconstruction of multivariate trigonometric polynomials with frequencies supported on arbitrary index sets of finite cardinality is considered, where rank-1 lattices are used as spatial discretizations. The approximation of multivariate smooth periodic functions by trigonometric polynomials is studied, based on a one-dimensional FFT applied to function samples. The smoothness of the functions is characterized via the decay of their Fourier coefficients, and various estimates for sampling errors are shown, complemented by numerical tests for up to 25 dimensions. In addition, the special case of perturbed rank-1 lattice nodes is considered, and a fast Taylor expansion based approximation method is developed. One main contribution is the transfer of the methods to the non-periodic case. Multivariate algebraic polynomials in Chebyshev form are used as ansatz functions and rank-1 Chebyshev lattices as spatial discretizations. This strategy allows for using fast algorithms based on a one-dimensional DCT. The smoothness of a function can be characterized via the decay of its Chebyshev coefficients. From this point of view, estimates for sampling errors are shown as well as numerical tests for up to 25 dimensions. A further main contribution is the development of a high-dimensional sparse FFT method based on rank-1 lattice sampling, which allows for determining unknown frequency locations belonging to the approximately largest Fourier or Chebyshev coefficients of a function. / In dieser Arbeit wird die schnelle Auswertung und Rekonstruktion multivariater trigonometrischer Polynome mit Frequenzen aus beliebigen Indexmengen endlicher Kardinalität betrachtet, wobei Rang-1-Gitter (rank-1 lattices) als Diskretisierung im Ortsbereich verwendet werden. Die Approximation multivariater glatter periodischer Funktionen durch trigonometrische Polynome wird untersucht, wobei Approximanten mittels einer eindimensionalen FFT (schnellen Fourier-Transformation) angewandt auf Funktionswerte ermittelt werden. Die Glattheit von Funktionen wird durch den Abfall ihrer Fourier-Koeffizienten charakterisiert und mehrere Abschätzungen für den Abtastfehler werden gezeigt, ergänzt durch numerische Tests für bis zu 25 Raumdimensionen. Zusätzlich wird der Spezialfall gestörter Rang-1-Gitter-Knoten betrachtet, und es wird eine schnelle Approximationsmethode basierend auf Taylorentwicklung vorgestellt. Ein wichtiger Beitrag dieser Arbeit ist die Übertragung der Methoden vom periodischen auf den nicht-periodischen Fall. Multivariate algebraische Polynome in Chebyshev-Form werden als Ansatzfunktionen verwendet und sogenannte Rang-1-Chebyshev-Gitter als Diskretisierungen im Ortsbereich. Diese Strategie ermöglicht die Verwendung schneller Algorithmen basierend auf einer eindimensionalen DCT (diskreten Kosinustransformation). Die Glattheit von Funktionen kann durch den Abfall ihrer Chebyshev-Koeffizienten charakterisiert werden. Unter diesem Gesichtspunkt werden Abschätzungen für Abtastfehler gezeigt sowie numerische Tests für bis zu 25 Raumdimensionen. Ein weiterer wichtiger Beitrag ist die Entwicklung einer Methode zur Berechnung einer hochdimensionalen dünnbesetzten FFT basierend auf Abtastwerten an Rang-1-Gittern, wobei diese Methode die Bestimmung unbekannter Frequenzen ermöglicht, welche zu den näherungsweise größten Fourier- oder Chebyshev-Koeffizienten einer Funktion gehören.

In a suggestion to build upon students’ historical knowledge through reading The Remains of the Day, this essay delves into cross-curricular teaching, dual-coded theory aspects and revision of suggested plans to improve learner’s understanding of historical novels, characters, events, and descriptions to grasp and reflect upon such historical knowledge. Several methods for enhancing students’ knowledge and to improve vocabulary knowledge are provided. The essay is theoretically based and presents different aspects of how students’ knowledge of English and history as separate subjects in school, could be combined to address the learning abilities of all students. Reading comprehension is mainly based upon students’ prior knowledge. Therefore, this essay delves into various parts of learners’ abilities to reflect upon the written word and its significance to reality. Furthermore, suggestions to how teachers can collaborate to achieve an improved understanding of the novel and its time period through history and vocabulary is presented.

Diploma thesis deals with the analysis of quality in digital television systems and contains theoretical description of subjective and objective assessment of quality picture methods. The thesis contains short-term and long-term analysis of quality picture of terrestrial television DVB-T. Measurements and experimentations were carried out with the help of Rohde&Schwarz DVQ analyzer of picture quality and software MPEG-2 Quality Monitor and MPEG-2 Elementary stream analyzer.

The first goal of diploma thesis is to study the basic principles of video signal compression. Introduction to techniques used to reduce irrelevancy and redundancy in the video signal. The second goal is, on the basis of information about compression tools, implement the individual compression tools in the programming environment of Matlab and assemble simple model of the video codec. Diploma thesis contains a description of the three basic blocks, namely - interframe coding, intraframe coding and coding with variable length word - according the standard MPEG-2.