Showing papers on "Fundamental frequency published in 2004"

Improved current control strategy for power conditioners using sinusoidal signal integrators in synchronous reference frame

Abstract: In this paper, a current control scheme, based on proportional-integral regulators using sinusoidal signal integrators (SSIs), is proposed for shunt type power conditioners. The aim is to simplify the implementation of SSI-based current harmonic compensation for industrial implementations where strict limitations on the harmonic distortion of the mains' currents are required. To compensate current harmonics, the SSIs are implemented to operate both on positive and negative sequence signals. One regulator, for the fundamental current component, is implemented in the stationary reference frame. The other regulators, for the current harmonics, are all implemented in a synchronous reference frame rotating at the fundamental frequency. This allows the simultaneous compensation of two current harmonics with just one regulator, yielding a significant reduction of the computational effort compared with other current control methods employing sinusoidal signal integrators implemented in stationary reference frame. A simple and robust voltage filter is also proposed by the authors to obtain a smooth and accurate position estimation of the voltage vector at the point of common coupling (PCC) under distorted mains' voltages. The whole control algorithm has been implemented on a 16-b, fixed-point digital signal processor (DSP) platform controlling a 20-kVA power conditioner prototype. The experimental results presented in this paper for inductive and capacitive loads show the validity of the proposed solutions.

Correct tonotopic representation is necessary for complex pitch perception

Abstract: The ability to extract a pitch from complex harmonic sounds, such as human speech, animal vocalizations, and musical instruments, is a fundamental attribute of hearing. Some theories of pitch rely on the frequency-to-place mapping, or tonotopy, in the inner ear (cochlea), but most current models are based solely on the relative timing of spikes in the auditory nerve. So far, it has proved to be difficult to distinguish between these two possible representations, primarily because temporal and place information usually covary in the cochlea. In this study, “transposed stimuli” were used to dissociate temporal from place information. By presenting the temporal information of low-frequency sinusoids to locations in the cochlea tuned to high frequencies, we found that human subjects displayed poor pitch perception for single tones. More importantly, none of the subjects was able to extract the fundamental frequency from multiple low-frequency harmonics presented to high-frequency regions of the cochlea. The experiments demonstrate that tonotopic representation is crucial to complex pitch perception and provide a new tool in the search for the neural basis of pitch.

Signal Processing Methods for the Automatic Transcription of Music

Abstract: Signal processing methods for the automatic transcription of music are developed in this thesis. Music transcription is here understood as the process of analyzing a music signal so as to write down the parameters of the sounds that occur in it. The applied notation can be the traditional musical notation or any symbolic representation which gives sufficient information for performing the piece using the available musical instruments. Recovering the musical notation automatically for a given acoustic signal allows musicians to reproduce and modify the original performance. Another principal application is structured audio coding: a MIDI-like representation is extremely compact yet retains the identifiability and characteristics of a piece of music to an important degree. The scope of this thesis is in the automatic transcription of the harmonic and melodic parts of real-world music signals. Detecting or labeling the sounds of percussive instruments (drums) is not attempted, although the presence of these is allowed in the target signals. Algorithms are proposed that address two distinct subproblems of music transcription. The main part of the thesis is dedicated to multiple fundamental frequency (F0) estimation, that is, estimation of the F0s of several concurrent musical sounds. The other subproblem addressed is musical meter estimation. This has to do with rhythmic aspects of music and refers to the estimation of the regular pattern of strong and weak beats in a piece of music. For multiple-F0 estimation, two different algorithms are proposed. Both methods are based on an iterative approach, where the F0 of the most prominent sound is estimated, the sound is cancelled from the mixture, and the process is repeated for the residual. The first method is derived in a pragmatic manner and is based on the acoustic properties of musical sound mixtures. For the estimation stage, an algorithm is proposed which utilizes the frequency relationships of simultaneous spectral components, without assuming ideal harmonicity. For the cancelling stage, a new processing principle, spectral smoothness, is proposed as an efficient new mechanism for separating the detected sounds from the mixture signal. The other method is derived from known properties of the human auditory system. More specifically, it is assumed that the peripheral parts of hearing can be modelled by a bank of bandpass filters, followed by half-wave rectification and compression of the subband signals. It is shown that this basic structure allows the combined use of time-domain periodicity and frequency-domain periodicity for F0 extraction. In the derived algorithm, the higher-order (unresolved) harmonic partials of a sound are processed collectively, without the need to detect or estimate individual partials. This has the consequence that the method works reasonably accurately for short analysis frames. Computational efficiency of the method is based on calculating a frequency-domain approximation of the summary autocorrelation function, a physiologically-motivated representation of sound. Both of the proposed multiple-F0 estimation methods operate within a single time frame and arrive at approximately the same error rates. However, the auditorily-motivated method is superior in short analysis frames. On the other hand, the pragmatically-oriented method is “complete” in the sense that it includes mechanisms for suppressing additive noise (drums) and for estimating the number of concurrent sounds in the analyzed signal. In musical interval and chord identification tasks, both algorithms outperformed the average of ten trained musicians.

Robust and accurate fundamental frequency estimation based on dominant harmonic components

Abstract: This paper presents a new method for robust and accurate fundamental frequency (F0) estimation in the presence of background noise and spectral distortion. Degree of dominance and dominance spectrum are defined based on instantaneous frequencies. The degree of dominance allows one to evaluate the magnitude of individual harmonic components of the speech signals relative to background noise while reducing the influence of spectral distortion. The fundamental frequency is more accurately estimated from reliable harmonic components which are easy to select given the dominance spectra. Experiments are performed using white and babble background noise with and without spectral distortion as produced by a SRAEN filter. The results show that the present method is better than previously reported methods in terms of both gross and fine F0 errors.

Accurate frequency estimation for real harmonic sinusoids

Abstract: A linear prediction based method is proposed for real harmonic sinusoidal frequency estimation. The estimator basically involves two steps. An initial fundamental frequency estimate is first obtained by solving a standard least-squares equation with exploitation of the harmonic structure of the sinusoidal signal or by using the MUSIC approach. Based on the initial estimate, an optimally weighted least squares cost function is then constructed from which the final estimate is acquired. Computer simulations show that the performance of the estimator approaches Crame/spl acute/r-Rao lower bound for sufficiently high signal-to-noise ratios and/or data lengths.

Relative contributions of temporal and place pitch cues to fundamental frequency discrimination in cochlear implantees.

Abstract: The effect of the filter bank on fundamental frequency (F0) discrimination was examined in four Nucleus CI24 cochlear implant subjects for synthetic stylized vowel-like stimuli. The four tested filter banks differed in cutoff frequencies, amount of overlap between filters, and shape of the filters. To assess the effects of temporal pitch cues on F0 discrimination, temporal fluctuations were removed above 10 Hz in one condition and above 200 Hz in another. Results indicate that F0 discrimination based upon place pitch cues is possible, but just-noticeable differences exceed 1 octave or more depending on the filter bank used. Increasing the frequency resolution in the F0 range improves the F0 discrimination based upon place pitch cues. The results of F0 discrimination based upon place pitch agree with a model that compares the centroids of the electrical excitation pattern. The addition of temporal fluctuations up to 200 Hz significantly improves F0 discrimination. Just-noticeable differences using both place and temporal pitch cues range from 6% to 60%. Filter banks that do not resolve the higher harmonics provided the best temporal pitch cues, because temporal pitch cues are clearest when the fluctuation on all channels is at F0 and preferably in phase.

Array measurements of S‐wave velocities from ambient vibrations

Abstract: SUMMARY The S-wave velocity is a very important factor in local hazard assessment. Direct measurement with conventional methods is very costly and therefore inexpensive and efficient methods are needed to make local hazard assessment more feasible. Techniques based on the analysis of recordings of ambient vibrations from small-scale arrays of sensors have become popular recently. One technique that is favoured by several research groups is the extraction of the dispersion curve by estimation of the f ‐k spectrum and its inversion for the S-wave velocity structure. This paper presents the results from an application based on high-resolution beamforming applied to the vertical component of the measurements. Synthetic ambient vibrations generated with a 2-D finite-difference code are used to illustrate and test the application. By superposition of random signals it is shown that the dispersion curve can be extracted even if 60 per cent of the wavefield consists of spatially uncorrelated signals. The errors in the phase velocities amount to less than 10 per cent. The dispersion curve can be extracted from the fundamental frequency of resonance upwards. Data from two real measurements are presented—from one site close to a city and another site within an industrial complex. The inverted S-wave velocity structures agree with reference data for the sites. The rule of thumb for the resolution of the method is confirmed.

Better place-coding of the fundamental frequency in cochlear implants

Abstract: In current cochlear implant systems, the fundamental frequency F0 of a complex sound is encoded by temporal fluctuations in the envelope of the electrical signals presented on the electrodes. In normal hearing, the lower harmonics of a complex sound are resolved, in contrast with a cochlear implant system. In the present study, it is investigated whether "place-coding" of the first harmonic improves the ability of an implantee to discriminate complex sounds with different fundamental frequencies. Therefore, a new filter bank was constructed, for which the first harmonic is always resolved in two adjacent filters, and the balance between both filter outputs is directly related to the frequency of the first harmonic. The new filter bank was compared with a filter bank that is typically used in clinical processors, both with and without the presence of temporal cues in the stimuli. Four users of the LAURA cochlear implant participated in a pitch discrimination task to determine detection thresholds for F0 differences. The results show that these thresholds decrease noticeably for the new filter bank, if no temporal cues are present in the stimuli. If temporal cues are included, the differences between the results for both filter banks become smaller, but a clear advantage is still observed for the new filter bank. This demonstrates the feasibility of using place-coding for the fundamental frequency.

The natural frequencies of the arterial system and their relation to the heart rate

Abstract: We assume the major function of the arterial system is transporting energy via its transverse vibration to facilitate the blood flowing all the way down to the microcirculation. A highly efficient system is related to maintaining a large pressure pulse along the artery for a given ventricular power. The arterial system is described as a composition of many infinitesimal Windkessels. The strong tethering in the longitudinal direction connects all the Windkessels together and makes them vibrate in coupled modes. It was assumed that at rest condition, the arterial system is in a steady distributed oscillatory state, which is the superposition of many harmonic modes of the transverse vibration in the arterial wall and the adherent blood. Every vibration mode has its own characteristic frequency, which depends on the geometry, the mass density, the elasticity, and the tethering of the arterial system. If the heart rate is near the fundamental natural frequency, the system is in a good resonance condition, we call this "frequency matching". In this condition, the pulsatile pressure wave is maximized. A pressure wave equation derived previously was used to predict this fundamental frequency. The theoretical result gave that heart rate is proportional to the average high-frequency phase velocity of the pressure wave and the inverse of the animal body length dimension. The area compliance related to the efficiency of the circulatory system is also mentioned.

Magnetic noise reduction method for AC rotary electric machine, and motor control apparatus and AC rotary electric machine apparatus using the same

Abstract: The invention provides techniques for reducing or altering the magnetic noise of an AC rotary electric machine. A magnetic noise reducing harmonic current of order n, whose frequency is n times the frequency of the fundamental frequency component of a polyphase AC current fed to an armature of a polyphase AC rotary electric machine, is superimposed on the polyphase AC current, thereby reducing or altering a harmonic component having a frequency (n−1) times the frequency of the fundamental frequency component and occurring due to a radial magnetic excitation force acting radially on an iron core of the AC rotary electric machine. Magnetic noise is caused by a vibration whose energy is the sum of the circumferential and radial vibrations of the iron core occurring due to the magnetomotive force of the rotor, and altering the radial vibration is particularly effective in altering the magnetic noise; as the harmonic component of the radially acting magnetic excitation force, occurring due to harmonic components having frequencies 3, 5, 7, and 13 times the fundamental frequency, has a frequency 6 or 12 times the fundamental frequency, the magnetomotive force of the rotor can be effectively reduced or altered when a current having a frequency 7 or 13 times the fundamental frequency is superimposed on the stator current.

Optimization of support positions to maximize the fundamental frequency of structures

Abstract: In this paper, the position optimization of simple supports is implemented to maximize the fundamental frequency of a beam or plate structure. Both elastic and rigid supports are taken into account. First, the frequency sensitivity with respect to the movement of a simple support is derived using the discrete method. By means of the shape functions of the finite element method, closed-form sensitivity formulations are developed straightforwardly. Then, a heuristic approach, called evolutionary shift method, is presented for optimizing support positions with a fixed grid mesh scheme. Based on the design sensitivity analysis, the support with the highest efficiency is shifted in priority along the elementary edges with the interval (step) of the elementary size. To facilitate the convergence of the process, the interpolation technique is employed to evaluate the solution more accurately. Finally, three numerical examples are presented to demonstrate the validity of the sensitivity analysis and the effectiveness of the optimization method. Copyright © 2004 John Wiley & Sons, Ltd.

On the Acoustic Sensitivity of a Symmetrical Two-Mass Model of the Vocal Folds to the Variation o fC ontrol Parameters

Abstract: Summary The acoustic properties of a recently proposed two-mass model for vocal-fold oscillations are analysed in terms of a set of acoustic parameters borrowed from phenomenological glottal-flow signal models. The analysed vocalfold model includes a novel description of flow separation within the glottal channel at a point whose position may vary in time when the channel adopts a divergent configuration. It also assumes a vertically symmetrical glottal structure, a hypothesis that does not hinder reproduction of glottal-flow signals and that reduces the number of control parameters of the dynamical system governing vocal-fold oscillations. Measuring the sensitivity of acoustic parameters to the variation of the model control parameters is essential to describe the actions that the modelled glottis employs to produce voiced sounds of different characteristics. In order to classify these actions, we applied an algorithmic procedure in which the implementation of the vocal-fold model is followed by a numerical measurement of the acoustic parameters describing the generated glottal-flow signal. We use this algorithm to generate a large database with the variation of acoustic parameters in terms of the model control parameters. We present results concerning fundamental frequency, intensity and pulse shape control in terms of subglottal pressure, muscular tension, and the effective mass of the folds participating in vocal-fold vibration. We also produce evidence for the identification o fv ocal-fold oscillatio nr egimes with the first and second laryngeal mechanisms, which are the most common phonation modes used in voiced-sound production. In terms of the model, the distinction between these mechanisms is closely related to the detection of glottal leakage, i.e. to an incomplete glottal closure during vocal-fold vibration. The algorithm is set to detect glottal leakage when transglottal air flow does not reach zero during the quasi-closed phase. I ti sa ls od esigned to simulate electroglottographic signals with the vocal-fold model. Numerical results are compared with experimental electroglottograms. In particular, a strong correspondence is found between the features of experimental and numerical electroglottograms during the transition between different laryngeal mechanisms.

High frequency voltage-fed inverter with phase-shift control for induction heating

Abstract: A voltage-fed resonant LCL inverter with phase-shift control is presented. The control strategy is seen to offer advantages in the megahertz operating region where a constant switching frequency is required. The inverter topology is inherently modular and higher output powers may therefore be readily achieved by adding additional MOSFET switching cells, the devices in each cell having only a modest rating. The inverter steady-state operation is analysed using fundamental frequency analysis. The predictions are verified through time-domain simulations and measure- ments of a 1.6 MHz 1 kW prototype.

Harmonic elimination for multilevel converter with programmed PWM method

Abstract: This work presents a programmed PWM method to eliminate specific higher order harmonics of multilevel converters. First, resultant theory is applied to transcendental equations to eliminate low order harmonics and to determine switching angles for the fundamental frequency switching techniques. Next the magnitudes and phases of the residual higher order harmonics are computed, generated, and subtracted from the original voltage waveform to eliminate these higher order harmonics. The simulation results show that the method can effectively eliminate the specific harmonics, and a low total harmonic distortion (THD) near sine wave is produced. An experimental 11-level H-bridge multilevel converter with a first-on, first-off switching strategy, which is used to balance loads between several levels, is employed to validate the method. The experimental results show that the method can effectively eliminate the specific harmonics, and the output voltage waveforms have less THD than that from the fundamental frequency switching techniques.

Speech information processing method and apparatus and storage medium using a segment pitch pattern model

Abstract: A speech information processing apparatus and method performs speech recognition. Speech is input, and feature parameters of the input speech are extracted. The feature parameters are recognized based on a segment pitch pattern model. The segment pitch pattern model may be obtained by modeling time change in a fundamental frequency of a phoneme belonging to a predetermined phonemic environment with a polynomial segment model. The segment pitch pattern model may also be obtained by modeling with at least one of a single mixed distribution and a multiple mixed distribution.

A linear frequency-domain model of a STATCOM

Abstract: A linear frequency-domain model of a static VAr compensator (STATCOM) is presented. Large signal transfers are derived to determine characteristic harmonics, and small-signal transfers are derived to determine noncharacteristic harmonics. Harmonic transfers are also derived for phase-locked loop and reactive power controls, and the response of a controlled STATCOM to a negative sequence fundamental frequency imbalance is demonstrated. As the model is linear, it is very computationally efficient, and is well suited to analysis of large networks with a number of similar connected FACTs devices. The importance of including full control dynamics for harmonic analysis is also clearly shown.

Ambient noise energy bursts observation and modeling: Trapping of harmonic structure-soil induced waves in a topmost sedimentary layer

Abstract: We study the nature of energy bursts that appeared in the frequency range 3–5 Hz in ambient seismic noise recorded in the Grenoble basin (French Alps) during a seismological array experiment. A close agreement is found between the identified azimuths of such noise bursts with the location of an industrial chimney. In-situ measurements of the chimney dynamic characteristics show a coincidence between the frequency of the first harmonic mode of the chimney and the fundamental frequency of a thin surficial layer that overlay the deep sediment fill. The interaction between the chimney and the surficial layer is then numerically simulated using simple impedance models and two soil profiles. Simulations exhibit a satisfactory agreement with observations and suggest that energy bursts result of inertial structure-soil interaction favored by resonance effects between the first harmonic mode of the structure and the fundamental frequency of the topmost layer.

Specmurt anasylis: a piano-roll-visualization of polyphonic music signal by deconvolution of log-frequency spectrum.

Abstract: In this paper, we propose a new signal processing technique, “specmurt anasylis,” that provides piano-rolllike visual display of multi-tone signals (e.g., polyphonic music). Specmurt is defined as inverse Fourier transform of linear spectrum with logarithmic frequency, unlike familiar cepstrum defined as inverse Fourier transform of logarithmic spectrum with linear frequency. We apply this technique to music signals frencyque anasylis using specmurt filreting instead of quefrency alanysis using cepstrum liftering. Suppose that each sound contained in the multi-pitch signal has exactly the same harmonic structure pattern (i.e., the energy ratio of harmonic components), in logarithmic frequency domain the overall shape of the multi-pitch spectrum is a superposition of the common spectral patterns with different degrees of parallel shift. The overall shape can be expressed as a convolution of a fundamental frequency pattern (degrees of parallel shift and power) and the common harmonic structure pattern. The fundamental frequency pattern is restored by division of the inverse Fourier transform of a given log-frequency spectrum, i.e., specmurt, by that of the common harmonic structure pattern. The proposed method was successfully tested on several pieces of music recordings.

Zero vector modulation method for voltage source inverter operating near zero output frequency

Abstract: A new zero vector modulation (ZVM) method is developed for usage in voltage source inverters operating near zero output frequency. In automotive applications during hill-holding maneuver a condition occurs requiring the electric drive system to produce high output current at zero or near zero output frequency. This mode of operation represents an increased stress for the VSI, since, depending upon the chosen modulation strategy, individual switch losses can exceed those occurring in normal operating modes. In the proposed ZVM method, the additional degree of freedom, choice of zero vectors, is used to distribute conduction losses among the inverter switches in the leg carrying the largest current. The zero vectors are altered periodically, at relatively low frequency (10 Hz-100 Hz). The method utilizes the fact that near zero speed, the average output voltage is small, and the time for which a zero vector is applied is significant. The method was tested experimentally on a 600 V, 400 A/sub rms/ inverter equipped with thermal-couples for the junction temperature measurements. For the tested inverter, the ZVM method enabled increased current rating at zero output fundamental frequency compared to commonly use discontinuous PWM method by approximately 15% and by about 30% compared to the space vector PWM modulation method with the same switching frequency.

Subspace-based fundamental frequency estimation

Abstract: In this paper, we present a subspace-based fundamental frequency estimator based on an extension of the MUSIC spectral estimator. A noise subspace is obtained from the eigenvalue decomposition of the estimated sample covariance matrix and fundamental frequency candidates are selected as the frequencies where the harmonic signal subspace is closest to being orthogonal to the noise subspace. The performance of the proposed method is evaluated and compared to that of the non-linear least-squares (NLS) estimator and the corresponding Cramer-Rao bound; it is concluded that the proposed method has good statistical performance at a lower computational cost than the statistically efficient NLS estimator.

Power quality indicators estimation using robust Newton-type algorithm

Abstract: A two-stage robust Newton-type numerical algorithm for power quality assessment in electric power systems is described. The robust Newton-type algorithm is applied to estimate current and voltage spectra and the fundamental frequency simultaneously, in the first algorithm stage. The algorithm robustness is achieved by developing an extra module called a 'bad data detector'. In the second algorithm stage power quality indicators are calculated, particularly the active, reactive, apparent and distortion powers. The main advantage is that the technique provides estimates which are insensitive to frequency deviations and to bad data appearing as a consequence of communication error, incomplete measurement, errors in mathematical models etc. The algorithm performance is tested under laboratory conditions using the distorted voltage and current signals digitised during an AC motor start. A simulation example of processing distorted currents and voltages of an arc furnace is also presented.

A Performance Enhancement Scheme for Robust Repetitive Control System

Abstract: This paper presents a method for enhancing the performance of a digital repetitive control system. The performance at the fundamental frequency and its harmonics of repetitive exogenous signals is improved by applying a modified low-pass q (z, z -1 ) filter structure in the repetitive signal generator of the internal model. Stability and robust performance is achieved through μ-synthesis. The modified low-pass filter is motivated by the attempt to reduce the sensitivity function by squaring it. The magnitude of the sensitivity function is substantially reduced at the periodic reference frequencies since the value is already much less than unity. The proposed approach is demonstrated by its implementation on an electrohydraulic actuator for tracking periodic cam-like profiles.

A new balanced harmonic load flow including nonlinear loads modeled with RBF networks

Abstract: In this paper, a steady-state frequency-domain model of an uncontrolled rectifier bridge with capacitive dc smoothing, developed with radial basis function (RBF) networks, is presented. The nonlinear load model is incorporated in a balanced harmonic load-flow program in order to evaluate the harmonic distortion introduced in the networks by this device. The fundamental frequency load flow, used in the harmonic load-flow algorithm, is enhanced with the incorporation to the Jacobian matrix of active and reactive power sensitivities from nonlinear loads. These sensitivities can be computed easily from the nonlinear load model built with RBF networks.