Audio Analysis/synthesis System

A method and apparatus for the automatic analysis, synthesis and modification of audio signals, based on an overlap-add sinusoidal model, is disclosed. Automatic analysis of amplitude, frequency and phase parameters of the model is achieved using an analysis-by-synthesis procedure which incorporates successive approximation, yielding synthetic waveforms which are very good approximations to the original waveforms and are perceptually identical to the original sounds. A generalized overlap-add sinusoidal model is introduced which can modify audio signals without objectionable artifacts. In addition, a new approach to pitch-scale modification allows for the use of arbitrary spectral envelope estimates and addresses the problems of high-frequency loss and noise amplification encountered with prior art methods. The overlap-add synthesis method provides the ability to synthesize sounds with computational efficiency rivaling that of synthesis using the discrete short-time Fourier transform (DSTFT) while eliminating the modification artifacts associated with that method.Georgia Tech Research Corporatio

Efficient algorithm for solving semi-infinite programming problems and their applications to nonuniform filter bank designs

An efficient algorithm for solving semi-infinite programming problems is proposed in this paper. The index set is constructed by adding only one of the most violated points in a refined set of grid points. By applying this algorithm for solving the optimum nonuniform symmetric/antisymmetric linear phase finite-impulse-response (FIR) filter bank design problems, the time required to obtain a globally optimal solution is much reduced compared with that of the previous proposed algorith

Techniques for Wideband All Digital Polar Transmission

abstract: Modern Communication systems are progressively moving towards all-digital transmitters (ADTs) due to their high efficiency and potentially large frequency range. While significant work has been done on individual blocks within the ADT, there are few to no full systems designs at this point in time. The goal of this work is to provide a set of multiple novel block architectures which will allow for greater cohesion between the various ADT blocks. Furthermore, the design of these architectures are expected to focus on the practicalities of system design, such as regulatory compliance, which here to date has largely been neglected by the academic community. Amongst these techniques are a novel upconverted phase modulation, polyphase harmonic cancellation, and process voltage and temperature (PVT) invariant Delta Sigma phase interpolation. It will be shown in this work that the implementation of the aforementioned architectures allows ADTs to be designed with state of the art size, power, and accuracy levels, all while maintaining PVT insensitivity. Due to the significant performance enhancement over previously published works, this work presents the first feasible ADT architecture suitable for widespread commercial deployment.Dissertation/ThesisDoctoral Dissertation Electrical Engineering 201

A highly efficient receiver for satellite-based Automatic Identification System signal detection

An innovative receiver architecture for the satellitebased Automatic Identification System (AIS) has been recently proposed. In this paper, we describe a few modifications that can be introduced on the algorithms for synchronization and detection, that provide an impressive performance improvement with respect to the previous system. The receiver architecture has been designed for an on-board implementation, and for this reason all algorithms have been realized keeping the complexity as low as possible. A prototype for the proposed receiver has been implemented by the University of Parma and CGS S.p.A. Compagnia Generale per 10 Spazio under the ESA project FENICE

Pilot patterns and power loading in NC-OFDM cognitive radios

Includes abstract.Includes bibliographical references.The implementation of cognitive radios is widely proposed through the use of Orthogonal Frequency Division Multiplexing (OFDM) modulation. In the special case of cognitive radios however, the OFDM modulation scheme cannot simply be implemented without modification due to the huge change in the basic laws of the transmission paradigm. The main reason behind this is that the modulation scheme can no longer assume the contiguousness of its band as well as the interference that may be caused by the cognitive radio users operating in such close proximity to the licensed users. The research presented in this thesis namely identified two areas of cognitive radio which addressed these issues. These were the power loading and channel estimation areas

Modelling and extraction of fundamental frequency in speech signals

This thesis was submitted for the degree of Doctor of Philosophy and awarded by Brunel University.One of the most important parameters of speech is the fundamental frequency of vibration of voiced sounds. The audio sensation of the fundamental frequency is known as the pitch. Depending on the tonal/non-tonal category of language, the fundamental frequency conveys intonation, pragmatics and meaning. In addition the fundamental frequency and intonation carry speaker gender, age, identity, speaking style and emotional state. Accurate estimation of the fundamental frequency is critically important for functioning of speech processing applications such as speech coding, speech recognition, speech synthesis and voice morphing. This thesis makes contributions to the development of accurate pitch estimation research in three distinct ways: (1) an investigation of the impact of the window length on pitch estimation error, (2) an investigation of the use of the higher order moments and (3) an investigation of an analysis-synthesis method for selection of the best pitch value among N proposed candidates. Experimental evaluations show that the length of the speech window has a major impact on the accuracy of pitch estimation. Depending on the similarity criteria and the order of the statistical moment a window length of 37 to 80 ms gives the least error. In order to avoid excessive delay as a consequence of using a longer window, a method is proposed ii where the current short window is concatenated with the previous frames to form a longer signal window for pitch extraction. The use of second order and higher order moments, and the magnitude difference function, as the similarity criteria were explored and compared. A novel method of calculation of moments is introduced where the signal is split, i.e. rectified, into positive and negative valued samples. The moments for the positive and negative parts of the signal are computed separately and combined. The new method of calculation of moments from positive and negative parts and the higher order criteria provide competitive results. A challenging issue in pitch estimation is the determination of the best candidate from N extrema of the similarity criteria. The analysis-synthesis method proposed in this thesis selects the pitch candidate that provides the best reproduction (synthesis) of the harmonic spectrum of the original speech. The synthesis method must be such that the distortion increases with the increasing error in the estimate of the fundamental frequency. To this end a new method of spectral synthesis is proposed using an estimate of the spectral envelop and harmonically spaced asymmetric Gaussian pulses as excitation. The N-best method provides consistent reduction in pitch estimation error. The methods described in this thesis result in a significant improvement in the pitch accuracy and outperform the benchmark YIN method

Design, manufacture and test of a magnetic encoder

An new eddy current based magnetic position encoder structure is proposed and studied in this thesis. The encoder is composed of one read head and one scale with metal plates placed periodically on a substrate. The read head contains one emitter and two receiver pairs which are all rectangular planar coils. The electromagnetic coupling between the emitter and receivers were affected by the relative position of the scale. A system level analytical model of the proposed encoder structure has been derived, from which three different encoder signals forms were generated. An amplification and synchronous demodulation circuit has been designed and fabricated. The circuit board was used successfully to process the encoder output signals in the measurement. Four PCB encoder prototypes were fabricated. These encoder structures were studied using the ANSYS MaxwellTM software package. The simulated and measured results were compared. The best accuracy performance of the PCB encoder is -15 μm to 15 μm from the simulation results and -35 μm to 25 μm from the corresponding measurement. An alternative manufacturing process of the magnetic encoder based on multilayer Low Temperature Co-fired Ceramic (LTCC) technology has also been presented. The fabrication process of the LTCC encoder and equipment used were described. Two different methods were used to characterise the LTCC encoder with good agreement between all approaches attempted. The best accuracy performance of the LTCC encoder was -30 μm to 25 μm and after lookup table correction the improved accuracy ranged from -10 μm to 10 μm