Speech has been the subject of interest for a very long time. Even with so much advancement in the processing techniques and in the understanding of the source of speech, it is, even today, rather difficult to generate speech in the laboratory in all its aspects. A simple aspect like how the speech can retain its intelligibility even if it is distorted or band passed is not really understood. This thesis deals with one small feature of speech viz., the intelligibility of speech is retained even when it is bandpassed with a minimum bandwidth of around 1 KHz located any where on the speech spectrum of 0-4 KHz. Several experiments have been conducted by the earlier workers by passing speech through various distortors like differentiators, integrators and infinite peak clippers and it is found that the intelligibility is retained to a very large extent in the distorted speech. The integrator and the differentiator remove essentially a certain portion of the spectrum. Therefore, it is thought that the intelligibility of the speech is spread over the entire speech spectrum and that, the intelligibility of speech may not be impaired even when it is bandpassed with a minimum bandwidth and the band may be located any where in the speech spectrum. To test this idea and establish this feature if it exists, preliminary experiments have been conducted by passing the speech through different filters and it is found that the conjecture seems to be on the right line. To carry out systematic experiments on this an experimental set up has been designed and fabricated which consists of a microprocessor controlled speech recording, storing and speech playback system. Also, a personal computer is coupled to the microprocessor system to enable the storage and processing of the data. Thirty persons drawn from different walks of life like teachers, mechanics and students have been involved for collecting the samples and for recognition of the information of the processed speech. Even though the sentences like 'This is devices lab' are used to ascertain the effect of bandwidth on the intelligibility, for the purpose of analysis, vowels are used as the speech samples. The experiments essentially consist of recording words and sentences spoken by the 30 participants and these recorded speech samples are passed through different filters with different bandwidths and central frequencies. The filtered output is played back to the various listeners and observations regarding the intelligibility of the speech are noted. The listeners do not have any prior information about the content of the speech. It has been found that in almost all (95%) cases, the messages or words are intelligible for most of the listeners when the band width of the filter is about 1 KHz and this is independent of the location of the pass band in the spectrum of 0-4 KHz. To understand how this feature of speech arises, spectrums of vowels spoken by 30 people have using FFT algorithms on the digitized samples of the speech. It is felt that there is a cyclic behavior of the spectrum in all the samples. To make sure that the periodicity is present and also to arrive at the periodicity, a moving average procedure is employed to smoothen the spectrum. The smoothened spectrums of all the vowels indeed show a periodicity of about 1 KHz. When the periodicities are analysed the average value of the periodicities has been found to be 1038 Hz with a standard deviation of 19 Hz. In view of this it is thought that the acoustic source responsible for speech must have generated this periodic spectrum, which might have been modified periodically to imprint the intelligibility. If this is true, one can perhaps easily understand this feature of the speech viz., the intelligibility is retained in a bandpassed speech of bandwidth 1 K H z . the pass band located any where in the speech spectrum of 0-4 KHz. This thesis describing the experiments and the analysis of the speech has been presented in 5 chapters. Chapter 1 deals with the basics of speech and the processing tools used to analyse the speech signal. Chapter 2 presents the literature survey from where the present problem is tracked down. Chapter 3 describes the details of the structure and the fabrication of the experimental setup that has been used. In chapter 4, the detailed account of the way in which the experiments are conducted and the way in which the speech is analysed is given. In conclusion in chapter 5, the work is summarised and the future work needed to establish the mechanism of speech responsible for the feature of speech described in this thesis is suggested.

Abstract Cognitive Radio (CR) improves the efficiency of spectrum utilization by allowing non- licensed users to utilize bands when not occupied by licensed users. In this thesis, we address several challenges currently limiting the wide use of cognitive radios. These challenges include identification of unoccupied bands, energy consumption and other technical challenges. Improved accuracy edge detection techniques are developed for CR to mitigate both noise and estimation error variance effects. Next, a reduced complexity Simplified DFT (SDFT) is proposed for use in CR. Then, a sub-Nyquist rate A to D converter is introduced to reduce energy consumption. Finally, a novel multi-resolution PSD estimation based on expectation-maximization algorithm is introduced that can obtain a more accurate PSD within a specified sensing time.

Abstract Cognitive Radio (CR) improves the efficiency of spectrum utilization by allowing non- licensed users to utilize bands when not occupied by licensed users. In this thesis, we address several challenges currently limiting the wide use of cognitive radios. These challenges include identification of unoccupied bands, energy consumption and other technical challenges. Improved accuracy edge detection techniques are developed for CR to mitigate both noise and estimation error variance effects. Next, a reduced complexity Simplified DFT (SDFT) is proposed for use in CR. Then, a sub-Nyquist rate A to D converter is introduced to reduce energy consumption. Finally, a novel multi-resolution PSD estimation based on expectation-maximization algorithm is introduced that can obtain a more accurate PSD within a specified sensing time.

Matched frequency responses are a fundamental starting point for a variety ofimplementations for microphone arrays. In this report, two methods for frequencyresponse-calibration of a pre-assembled microphone array are presented andevaluated. This is done by extracting the deviation in frequency responses of themicrophones in relation to a selected reference microphone, using a swept sine asa stimulus signal and an inverse filter. The swept sine includes all frequencieswithin the bandwidth of human speech. This allows for a full frequency responsemeasurements from all microphones using a single recording.Using the swept sine, the deviation in frequency response between the microphonescan be obtained. This deviation represents the scaling factor that all microphonesmust be calibrated with to match the reference microphone. Applying the scalingfactors on the recorded stimulus signal shows an improvement for both implementedmethods, and where one method matches the frequency response of the microphoneswith high accuracy.Once the scaling factors of the various microphones is obtained, it can be usedto calibrate other recorded signals. This leads to an minor improvement formatching the frequency responses, as it has been shown that the differencesin frequency response between the microphones is signal-dependent and variesbetween recordings. The response differences between the microphones dependson the design of the array, speaker, room and the acoustic frequency dispersionthat occurs with sound waves. This makes it difficult to calibrate the frequencyresponses of the microphones without appropriate equipment because the responseof the microphones is noticeably affected by these other factors. Proposals to addressthese problems are discussed in the report as future work. / Matchade frekvenssvar är en grundläggande utgångspunkt för ett flertal implementationer för mikrofonarrayer. I denna rapport presenteras och utvärderas tvåmetoder för frekvenssvarskalibrering för en förmonterad mikrofonarray. Detta görsgenom att extrahera avvikelsen i frekvenssvar hos alla mikrofoner i förhållandetill en vald referensmikrofon. Frekvenssvaren tas fram med hjälp av ettsinussvep som stimulanssignal och ett inverterat filter. Sinussvepet inkluderar helafrekvensbredden för mänskligt tal och möjliggör att mikrofonernas fulla frekvenssvarkan analyseras från en enda inspelning.Med hjälp av sinussvepet kan avvikelsen i frekvenssvar mellan mikrofonerna erhållas.Denna avvikelse representerar den skalningsfaktor alla mikrofoner måste kalibrerasefter för att matcha referensmikrofonen. Genom att applicera faktorerna på deninspelade stimulussignalen ses en förbättring för båda implementerade metoderna,där en metod matchar mikrofonernas frekvenssvar med hög noggrannhet.När skalningsfaktorn för de olika mikrofonerna har erhållits kan den användas föratt kalibrera andra inspelade signaler. Detta leder till en liten förbättring i att matchafrekvenssvaren, då det har visat sig att skillnader mellan mikrofonernas frekvenssvarär signalberoende och varierar mellan inspelningar. Skillnader i frekvenssvar mellanmikrofonerna beror på ljudets utbredning i rummet, utformningen av arrayen,högtalaren och den akustiska frekvensspridningen som uppstår hos ljudvågor. Dettagör det svårt att kalibrera frekvenssvaren hos mikrofonerna utan lämplig utrustningeftersom mikrofonernas respons märkbart påverkas av dessa andra faktorer. Förslagför att kringgå dessa problem diskuteras i rapporten och tas upp som framtidaarbete.

Processing of broadband signals induced on an antenna array using a tapped delay line filter and a set of steering delays has two problems. Firstly one needs to manipulate large matrices to estimate the filter coefficients. Secondly the use of steering delays is not only cumbersome but implementation errors cause loss of system performance. This thesis looks at both of these problems and presents elegant solutions by developing and studying a design method referred to as the DFT method, which does not require steering delays and is computationally less demanding compared to existing methods. Specifically the thesis studies and compares the performance of a time domain element space beamformer using the proposed method and that using an existing method, and develops the DFT method when the processor is implemented in partitioned form. The study presented in the thesis shows that the processors using the DFT method are robust to look direction errors and require less computation than that using the existing method for comparable performance. The thesis further introduces a broadband beamformer design which does not require any steering delays between the sensors and the tapped delay line section as is presently the case. It has the capability of steering the array in an arbitrary direction with a specified frequency response in the look direction while canceling unwanted uncorrelated interferences. The thesis presents and compares the performance of a number of techniques to synthesize an antenna pattern of a broadband array. These techniques are designed to produce isolated point nulls as well as broad sector nulls and to eliminate the need for the steering delays. Two of the pattern synthesis techniques presented in the thesis allow optimization against unwanted interferences in unknown directions. The techniques allow formulation of a beamforming problem such that the processor is not only able to place nulls in specified directions but also able to cancel directional interferences in unknown directions along with a specified frequency response in the look direction over a band of interest. The thesis also presents a set of directional constraints such that one does not need steering delays and an array can be constrained in an arbitrary direction with a specified frequency response. The constraints presented in the thesis are simple to implement. Based on these constraints a pattern synthesis technique for broadband antenna array is also presented.

Submitted by Renata Lopes (renatasil82@gmail.com) on 2017-04-25T17:30:25Z No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 10221613 bytes, checksum: 0d6eef0f715fc0f9f68bf12a390dcd55 (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2017-04-26T12:22:24Z (GMT) No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 10221613 bytes, checksum: 0d6eef0f715fc0f9f68bf12a390dcd55 (MD5) / Made available in DSpace on 2017-04-26T12:22:24Z (GMT). No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 10221613 bytes, checksum: 0d6eef0f715fc0f9f68bf12a390dcd55 (MD5) Previous issue date: 2015-03-26 / CAPES - Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Esta tese apresenta as etapas de desenvolvimento de um sistema de monitoramento de parâmentos de qualidade de energia dedicado a análise de harmônicos variantes no tempo através do equipamento denominado AHVT (Analisador de Harmônicos Variantes no Tempo). O desenvolvimento do trabalho é composto por: (i) estudo e implementação MATLAB de algoritmos para processamento em tempo real, com capacidade de sintonização dos componentes harmônicos; (ii) análise e desenvolvimento de estratégias para detecção e validação da presença de interharmônicos próximos à frequência fundamental e suas consequência na estimação de parâmetros como fase, amplitude e frequência para o componente fundamental, (iii) proposta de implementação do dispositivo, sistema de aquisição/ condicionamento de sinais/ filtragem, sistema de conversão analógico digital e plataforma de processamentoDSP/FPGA, sistema de transmissão de dados e plataformas de análise online/offline dos eventos de harmônicos variantes no tempo; (iv) plataforma de simulação do Analisador de Harmônicos Variantes no Tempo (AHVT) para estudo dos métodos de trigger para detecção e captura dos eventos. / In this work is presented the steps of development and implementation of a Power Quality paramaters monitoring system with main goal events denomined ”time arying harmonics”named of Time Varying Harmonic Analyzer. The development is comprises:(i) research and implementation of real time algorithms with capable to tuning harmonic waves,(ii) Analyze and research/development of strategies for detect and validation of interharmonics with frequencies near of fundamental, and conseguencies and challenges to phase, magnitude and frequency estimation with presence interharmonic waveform (iii) The proposal of a hardware design including analog to digital conversion and digital signal processing plataform, broadcast data link and IHM(Interface Human Machine) for online and offline analyzes to time varying harmonic analyzer;(iiii)off-line simulation plataform of Analisador de Harmônicos Variantes no Tempo Time Varying Harmonic Analyzer (TVHA) to trigger detect methods to detection and capture of waveforms.

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.

In large antenna systems, traditional channel estimation is costly and infeasible in some situations. Compressive sensing was proposed to estimate the channel with fewer measurements. Most of the previous work uses a predefined discrete Fourier transform matrix or overcomplete Fourier transform matrix to approximate the channel. Then, a learned dictionary trained by K-singular value decomposition (K-SVD) was proposed and was proved superiority using orthogonal matching pursuit (OMP) to reconstruct the sparse channel. However, with the development of compressive sensing, there are plenty of dictionary learning algorithms and sparse recovery algorithms. It is important to identify the effect and the performance of different algorithms when transforming the high dimensional channel vectors to low dimensional representations. In this thesis, we use a spatial channel model to generate channel vectors. Dictionaries are trained by K-SVD and method of optimal directions (MOD). Several sparse recovery algorithms are used to find the sparse approximation of the channel like OMP and gradient descent with sparsification (GraDeS). We present simulation results and discuss the performance of the various algorithms in terms of accuracy, sparsity, and complexity. We find that predefined dictionaries works with most of the algorithms in sparse recovery but learned dictionaries only work with pursuit algorithms, and only show superiority when the algorithm coincides with the algorithm in the sparse coding stage. / I stora antennsystem är traditionell kanaluppskattning kostsam och omöjlig i vissa situationer. Kompressionsavkänning föreslogs för att uppskatta kanalen med färre mätningar. Det mesta av det tidigare arbetet använder en fördefinierad diskret Fourier transformmatris eller överkompletterad Fourier -transformmatris för att approximera kanalen. Därefter föreslogs en inlärd ordbok som utbildats av K-SVD och bevisades överlägsen med hjälp av OMP för att rekonstruera den glesa kanalen. Men med utvecklingen av komprimerad avkänning finns det gott om algoritmer för inlärning av ordlistor och glesa återställningsalgoritmer. Det är viktigt att identifiera effekten och prestandan hos olika algoritmer när de högdimensionella kanalvektorerna omvandlas till lågdimensionella representationer. I denna avhandling använder vi en rumslig kanalmodell för att generera kanalvektorer. Ordböcker tränas av K-SVD och MOD. Flera glesa återställningsalgoritmer används för att hitta den glesa approximationen av kanalen som OMP och GraDeS. Vi presenterar simuleringsresultat och diskuterar prestanda för de olika algoritmerna när det gäller noggrannhet, sparsamhet och komplexitet. Vi finner att fördefinierade ordböcker fungerar med de flesta algoritmerna i gles återhämtning, men inlärda ordböcker fungerar bara med jaktalgoritmer och visar bara överlägsenhet när algoritmen sammanfaller med algoritmen i det glesa kodningsstadiet.

本文希望藉著K.T Arasu, D.Jungnickel, A.Pott推廣Wilbrink定理的方法去尋找Wilbrink等式的推廣式在p^k∥n,k≧4的推廣式和其應用。 / In this thesis we formulate and provide rigorous proofs of Wilbrink's theorem and it's variations due to Arasu, A.Pott and D.Jungnickel. some questions on further generalizations of Wilbrink's theorem are discussed; known generalization are study in A.Pott's dissertation.

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