Modern Methods of Time-Frequency Warping of Sound Signals

Tato práce se zabývá reprezentací nestacionárních harmonických signálů s časově proměnnými komponentami. Primárně je zaměřena na Harmonickou transformaci a jeji variantu se subkvadratickou výpočetní složitostí, Rychlou harmonickou transformaci. V této práci jsou prezentovány dva algoritmy využívající Rychlou harmonickou transformaci. Prvni používá jako metodu odhadu změny základního kmitočtu sbírané logaritmické spektrum a druhá používá metodu analýzy syntézou. Oba algoritmy jsou použity k analýze řečového segmentu pro porovnání vystupů. Nakonec je algoritmus využívající metody analýzy syntézou použit na reálné zvukové signály, aby bylo možné změřit zlepšení reprezentace kmitočtově modulovaných signálů za použití Harmonické transformace.This thesis deals with representation of non-stationary harmonic signals with time-varying components. Its main focus is aimed at Harmonic Transform and its variant with subquadratic computational complexity, the Fast Harmonic Transform. Two algorithms using the Fast Harmonic Transform are presented. The first uses the gathered log-spectrum as fundamental frequency change estimation method, the second uses analysis-by-synthesis approach. Both algorithms are used on a speech segment to compare its output. Further the analysis-by-synthesis algorithm is applied on several real sound signals to measure the increase in the ability to represent real frequency-modulated signals using the Harmonic Transform.

A Tutorial on Speech Synthesis Models

For Speech Synthesis, the understanding of the physical and mathematical models of speech is essential. Hence, Speech Modeling is a large field, and is well documented in literature. The aim in this paper is to provide a background review of several speech models used in speech synthesis, specifically the Source Filter Model, Linear Prediction Model, Sinusoidal Model, and Harmonic/Noise Model. The most important models of speech signals will be described starting from the earlier ones up until the last ones, in order to highlight major improvements over these models. It would be desirable a parametric model of speech, that is relatively simple, flexible, high quality, and robust in re-synthesis. Emphasis will be given in Harmonic / Noise Model, since it seems to be more promising and robust model of speech. (C) 2015 The Authors. Published by Elsevier B.V

Spectral analysis for nonstationary audio

A new approach for the analysis of nonstationary signals is proposed, with a focus on audio applications. Following earlier contributions, nonstationarity is modeled via stationarity-breaking operators acting on Gaussian stationary random signals. The focus is on time warping and amplitude modulation, and an approximate maximum-likelihood approach based on suitable approximations in the wavelet transform domain is developed. This paper provides theoretical analysis of the approximations, and introduces JEFAS, a corresponding estimation algorithm. The latter is tested and validated on synthetic as well as real audio signal.Comment: IEEE/ACM Transactions on Audio, Speech and Language Processing, Institute of Electrical and Electronics Engineers, In pres

Influence of parameters of modal analysis on vibration-based structural damage detectability

Vibration-based non-destructive testing (NDT) is one of the most widely used diagnostic methods of composite structures. The parameters of modal analysis may affect the accuracy of damage detectability, localization and identification. The aim of this paper is to investigate an influence of such parameters on results obtained after modal analysis of a composite structure and wavelet-based processing. Four parameters were taken into consideration: a frequency resolution of a frequency response function, a number of averaging cycles, a type of an excitation signal and a number of measurement points. The series of tests were performed on a composite sandwich structure with a honeycomb-type core using scanning laser Doppler vibrometer. The discussed results can be considered as recommendations for performing NDT of composite structures using vibrations in terms of parameters of modal analysis

Audio source separation techniques including novel time-frequency representation tools

The thesis explores the development of tools for audio representation with applications in Audio Source Separation and in the Music Information Retrieval (MIR) field. A novel constant Q transform was introduced, called IIR-CQT. The transform allows a flexible design and achieves low computational cost. Also, an independent development of the Fan Chirp Transform (FChT) with the focus on the representation of simultaneous sources is studied, which has several applications in the analysis of polyphonic music signals. Dierent applications are explored in the MIR field, some of them directly related with the low-level representation tools that were analyzed. One of these applications is the development of a visualization tool based in the FChT that proved to be useful for musicological analysis . The tool has been made available as an open source, freely available software. The proposed Transform has also been used to detect and track fundamental frequencies of harmonic sources in polyphonic music. Also, the information of the slope of the pitch was used to define a similarity measure between two harmonic components that are close in time. This measure helps to use clustering algorithms to track multiple sources in polyphonic music. Additionally, the FChT was used in the context of the Query by Humming application. One of the main limitations of such application is the construction of a search database. In this work, we propose an algorithm to automatically populate the database of an existing Query by Humming, with promising results. Finally, two audio source separation techniques are studied. The first one is the separation of harmonic signals based on the FChT. The second one is an application for which the fundamental frequency of the sources is assumed to be known (Score Informed Source Separation problem)

Superharmonic imaging with chirp coded excitation: Filtering spectrally overlapped harmonics

Superharmonic imaging improves the spatial resolution by using the higher order harmonics generated in tissue. The superharmonic component is formed by combining the third, fourth, and fifth harmonics, which have low energy content and therefore poor SNR. This study uses coded excitation to increase the excitation energy. The SNR improvement is achieved on the receiver side by performing pulse compression with harmonic matched filters. The use of coded signals also introduces new filtering capabilities that are not possible with pulsed excitation. This is especially important when using wideband signals. For narrowband signals, the spectral boundaries of the harmonics are clearly separated and thus easy to filter; however, the available imaging bandwidth is underused. Wideband excitation is preferable for harmonic imaging applications to preserve axial resolution, but it generates spectrally overlapping harmonics that are not possible to filter in time and frequency domains. After pulse compression, this overlap increases the range side lobes, which appear as imaging artifacts and reduce the Bmode image quality. In this study, the isolation of higher order harmonics was achieved in another domain by using the fan chirp transform (FChT). To show the effect of excitation bandwidth in superharmonic imaging, measurements were performed by using linear frequency modulated chirp excitation with varying bandwidths of 10% to 50%. Superharmonic imaging was performed on a wire phantom using a wideband chirp excitation. Results were presented with and without applying the FChT filtering technique by comparing the spatial resolution and side lobe levels. Wideband excitation signals achieved a better resolution as expected, however range side lobes as high as -23 dB were observed for the superharmonic component of chirp excitation with 50% fractional bandwidth. The proposed filtering technique achieved >50 dB range side lobe suppression and improved the image quality without affecting the axial resolution

Kolmio- ja ramppiaallot kohteen havaitsemisessa taajuusmoduloidulla kantoaaltotutkalla

The goal of this thesis was to study how a triangular chirp can be used in target detection and parameter estimation with an FMCW radar. The history of radar technology is briefly discussed, and motivation for the research is presented with a review of some FMCW radar applications. The triangular chirp is compared with slow-time processed ramp chirps on a theoretical basis. A method of improving the accuracy of triangular chirps with zero padding is presented. The process of zero padding is demonstrated with a MATLAB example, and then applied to real measurement data. The measurements were performed in an anechoic chamber and an office environment. Range and radial velocity of a single target were considered. A walking person was used as the test target. A highly accurate laser sensor was used as a reference. The results demonstrate that the accuracy of a triangular chirp can be greatly improved with zero padding, which allows much shorter chirps to be used while maintaining a high accuracy. For example, a zero padded 1000 μs long triangular chirp was used to determine the radial velocity of a walking target with an accuracy of approximately 0.25 m/s. In comparison, without zero padding the accuracy was approximately 2 m/s. To reach a comparable accuracy without zero padding, the triangular chirp would have to be significantly longer. At the end of the thesis, topics for further research are proposed.Työssä tutkitaan, kuinka kolmioaaltoa voidaan käyttää havaitun kohteen paikan ja nopeuden määrittämiseen FMCW-tutkalla. Käsittelen alussa lyhyesti tutkien historiaa, ja FMCW-tutkien yleisiä sovelluksia. Kolmioaaltoa verrataan ramppiaaltoon teoreettiselta pohjalta. Esittelen työssä menetelmän kolmioaallon mittaustarkkuuden parantamiseksi lisäämällä näytteistettyyn signaaliin nollia. Menetelmää demonstroidaan MATLAB:illa ja lopulta sovelletaan mitattuun dataan. Mittaukset suoritettiin radiokaiuttomassa huoneessa ja toimistotilassa. Tilaisuuksissa pyrittiin mittaamaan yksittäisen kohteen sijaintia ja liikenopeutta. Mittauskohteena toimi kävelevä ihminen. Mittausten vertailukohteena toimi lasersensori. Saadut tulokset demonstroivat kolmioaallon tarkkuuden olevan merkittävästi parannettavissa esitetyllä menetelmällä. Korkean tarkkuuden säilyttäminen on mahdollista lyhyelläkin kolmioaallolla, joka ilman nollien lisäämistä olisi erittäin epätarkka. Esimerkiksi kävelevän kohteen nopeus mitattiin 1000 μs pituisella kolmioaallolla ja nollien lisäämisellä noin 0.25 m/s tarkkuudella, kun taas ilman nollia tarkkuus oli noin 2 m/s. Ilman nollien lisäystä kolmioaallon pituuden olisi oltava moninkertainen vastaavan tarkkuuden saavuttamiseksi. Työn lopussa esitetään aiheita jatkotutkimukselle

Various modulated hybrid pulse compression for advanced ultrasound technology and its non-destructive testing application

This thesis was submitted for the degree of Doctor of Philosophy and awarded by Brunel University LondonUltrasound is a sound wave with a frequency greater than 20 kHz. It obeys the propagation laws of reflection, refraction, diffraction, and scattering. Because of its excellent physical properties, ultrasound has been used in a variety of fields, including industry and medicine. There are many techniques that use ultrasound as detection methods in the field of non-destructive testing (NDT) and medical treatment. In a typical ultrasound system, a sine wave or pulse signal with a fit window is considered as the transmitted signal. This results in low accuracy in some special situations, such as testing high attenuation material. The signal-to-noise ratio (SNR) is an important parameter for evaluating the performance of an echo signal or imaging. However, under high attenuation materials or noisy conditions, SNR will significantly decrease. Under these conditions, valid information in the received signal will be obscured by noise. This situation can cause errors in the detection system. In an ultrasound system, increasing the SNR of the echo signal can reduce detection errors and improve accuracy. First, in ultrasound systems, a noise reduction method based on pulse compression has been investigated and applied. Convolution and modulation were used in the proposed method to generate new hybrid emission signals. The hybrid codes can only be distinguished by a special matched filter that is related to the emission signals. The echo signals processed by a special matched filter have a high main lobe and a very low side lobe, implying that the side lobe level and SNR will increase. When compared to traditional denoising methods, the proposed method can significantly improve SNR while only requiring a change in the transmission code without requiring any hardware changes. Second, in a low voltage ultrasonic testing (UT) system, a hybrid phase modulated code excitation method based on the Barker and Golay code pairs was proposed and implemented. In a UT system, the lower the pulsing voltage, the lower the SNR of the signal. Attempting to reduce the pulsing voltage will result in noisy and unusable results. The proposed hybrid method can increase main lobe power in low average power transmitted and received signals. The proposed method has been theoretically examined and then tested in simulation studies. The experimental results showed that the main lobe level of the code produced by convolution of Barker code and Golay code pairs is around 30 dB higher than the simple pulse, and the main lobe of the combined code is around 15 dB higher than the traditional Barker code, with the sidelobe being the same as the Baker code that constitutes this combined code. As a result, the combined code’s peak sidelobe level (PSL) is approximately 5 dB lower than the traditional Barker code. Because of this, UT devices can be used in real-world applications, even in low-voltage situations. Third, the torsional wave mode T(0,1) hybrid phase modulated code excitation method has been proposed and applied in a long range guided wave testing (GWT) system. The proposed hybrid method combines the Barker and Golay code pair and is modulated by a fitted sine wave. This method combines the benefits of these two coding methods and increases code length flexibility. The SNR and PSL of the processed signal are used to assess the method’s performance. The proposed method has been tested in GWT using both finite element method (FEM) simulation and real-world testing. The results of pipeline laboratory testing revealed that the best increasing SNR of BCG is around 33.5 dB when compared to a simple pulse at 40 kHz, and the peak sidelobe level is around -24 dB. The proposed method, as well as other traditional methods, were used for pipeline defect detection testing. The results of the tests showed that the hybrid coded excitation method can detect notches that are difficult to detect with other methods and effectively improve the SNR. The applied method’s increasing SNR is around 6 dB, which agrees with the simulation and laboratory testing results. In UGW testing, the proposed coded excitation method was highly regarded. Finally, the non-linear frequency modulated (NLFM) hybrid pulse compression method has been proposed and implemented in an ultrasound imaging (UI) system. The proposed code combines the Barker and Golay codes and is modulated using a non-linear frequency method based on the Zak transform. Theoretical research on signal generation and decoding has been presented, as well as cyst phantom simulation. The simulation analysis shows that the novel code method can improve the contrast ratio by 15.96 dB and the SNR by 36.64 dB when compared to a simple pulse signal. Overall, this study demonstrated that the proposed novel method can be effectively used in ultrasound detection methods to improve performance

Acousto-optic systems for advanced microscopy

Acoustic waves in an optical medium cause rapid periodic changes in the refraction index, leading to diffraction effects. Such acoustically controlled diffraction can be used to modulate, deflect, and focus light at microsecond timescales, paving the way for advanced optical microscopy designs that feature unprecedented spatiotemporal resolution. In this article, we review the operational principles, optical properties, and recent applications of acousto-optic (AO) systems for advanced microscopy, including random-access scanning, ultrafast confocal and multiphoton imaging, and fast inertia-free light-sheet microscopy. As AO technology is reaching maturity, designing new microscope architectures that utilize AO elements is more attractive than ever, providing new exciting opportunities in fields as impactful as optical metrology, neuroscience, embryogenesis, and high-content screening