Download Real-Time Wave Digital Simulation of Cascaded Vacuum Tube Amplifiers using Modified Blockwise Method Vacuum tube amplifiers, known for their acclaimed distortion characteristics, are still widely used in hi-fi audio devices. However, bulky, fragile and power-consuming vacuum tube devices have also motivated much research on digital emulation of vacuum tube amplifier behaviors. Recent studies on Wave Digital Filters (WDF) have made possible the modeling of multi-stage vacuum tube amplifiers within single WDF SPQR trees. Our research combines the latest progress on WDF with the modified blockwise method to reduce the overall computational complexity of modeling cascaded vacuum tube amplifiers by decomposing the whole circuit into several small stages containing only two adjacent triodes. Certain performance optimization methods are discussed and applied in the eventual real-time implementation.
Download Joint modeling of impedance and radiation as a recursive parallel filter structure for efficient synthesis of wind instrument sound In the context of efficient synthesis of wind instrument sound, we introduce a technique for joint modeling of input impedance and sound pressure radiation as digital filters in parallel form, with the filter coefficients derived from experimental data. In a series of laboratory measurements taken on an alto saxophone, the input impedance and sound pressure radiation responses were obtained for each fingering. In a first analysis step, we iteratively minimize the error between the frequency response of an input impedance measurement and that of a digital impedance model constructed from a parallel filter structure akin to the discretization of a modal expansion. With the modal coefficients in hand, we propose a digital model for sound pressure radiation which relies on the same parallel structure, thus suitable for coefficient estimation via frequency-domain least-squares. For modeling the transition between fingering positions, we propose a simple model based on linear interpolation of input impedance and sound pressure radiation models. For efficient sound synthesis, the common impedance-radiation model is used to construct a joint reflectanceradiation digital filter realized as a digital waveguide termination that is interfaced to a reed model based on nonlinear scattering.
Download Contact Sensor Processing for Acoustic Instrument Sensor Matching Using a Modal Architecture This paper proposes a method to filter the output of instrument contact sensors to approximate the response of a well placed microphone. A modal approach is proposed in which mode frequencies and damping ratios are fit to the frequency response of the contact sensor, and the mode gains are then determined for both the contact sensor and the microphone. The mode frequencies and damping ratios are presumed to be associated with the resonances of the instrument. Accordingly, the corresponding contact sensor and microphone mode gains will account for the instrument radiation. The ratios between the contact sensor and microphone gains are then used to create a parallel bank of second-order biquad filters to filter the contact sensor signal to estimate the microphone signal.
Download FAST MUSIC – An Efficient Implementation Of The Music Algorithm For Frequency Estimation Of Approximately Periodic Signals Noise subspace methods are popular for estimating the parameters of complex sinusoids in the presence of uncorrelated noise and have applications in musical instrument modeling and microphone array processing. One such algorithm, MUSIC (Multiple Signal Classification) has been popular for its ability to resolve closely spaced sinusoids. However, the computational efficiency of MUSIC is relatively low, since it requires an explicit eigenvalue decomposition of an autocorrelation matrix, followed by a linear search over a large space. In this paper, we discuss methods for and the benefits of converting the Toeplitz structure of the autocorrelation matrix to circulant form, so that eigenvalue decomposition can be replaced by a Fast Fourier Transform (FFT) of one row of the matrix. This transformation requires modeling the signal as at least approximately periodic over some duration. For these periodic signals, the pseudospectrum calculation becomes trivial and the accuracy of the frequency estimates only depends on how well periodicity detection works. We derive a closed-form expression for the pseudospectrum, yielding large savings in computation time. We test our algorithm to resolve closely spaced piano partials.