Download Wave Digital Modeling of the Output Chain of a Vacuum-Tube Amplifier
This article introduces a physics-based real-time model of the output chain of a vacuum-tube amplifier. This output chain consists of a single-ended triode power amplifier stage, output transformer, and a loudspeaker. The simulation algorithm uses wave digital filters in digitizing the physical electric, mechanic, and acoustic subsystems. New simulation models for the output transformer and loudspeaker are presented. The resulting real-time model of the output chain allows any of the physical parameters of the system to be adjusted during run-time.
Download State-Space Representation for Digital Waveguide Networks of Lossy Flared Acoustic Pipes
This paper deals with digital waveguide modeling of wind instruments. It presents the application of state-space representations to the acoustic model of Webster-Lokshin. This acoustic model describes the propagation of longitudinal waves in axisymmetric acoustic pipes with a varying cross-section, visco-thermal losses at the walls, and without assuming planar or spherical waves. Moreover, three types of discontinuities of the shape can be taken into account (radius, slope and curvature), which can lead to a good fit of the original shape of pipe. The purpose of this work is to build low-cost digital simulations in the time domain, based on the Webster-Lokshin model. First, decomposing a resonator into independent elementary parts and isolating delay operators lead to a network of input/output systems and delays, of KellyLochbaum network type. Second, for a systematic assembling of elements, their state-space representations are derived in discrete time. Then, standard tools of automatic control are used to reduce the complexity of digital simulations in time domain. In order to validate the method, simulations are presented and compared with measurements.
Download Nonlinear Circuit Simulation using Time‐Variant Filter
The dynamic simulation of nonlinear guitar effects has recently been studied in depth. There are several approaches to the simulation of the distortion guitar effects. This paper presents an algorithm based on using a digital linear time-variant filter for the simulation of the nonlinear circuit of diode limiter. Filter coefficients are changed in each sample period according to the level of an input and output signal using numerical methods for the solution of nonlinear functions. The designed algorithm was used in the distortion effect to examine its characteristics. Sound examples of the implemented distortion effect can be found at web page
Download Asymmetries make the difference: A nonlinear model of transistor-based analog ring modulators
This work analyzes analog ring modulators based on bipolar transistors, such as the EMS VCS3 and the Doepfer A-114. It is shown that the perfectly symmetric standard model from literature [1][2] does not suffice to describe crucial first-order effects. A detailed analysis of the circuit using mismatched parts is performed. The insights gained from this analysis are used to formulate a digital model which can be easily implemented and which captures the essential audible effects.
Download Instantaneous Harmonic Analysis for Vocal Processing
The paper considers the application of instantaneous harmonic analysis to a real-time vocal processing system for pitch, timbre and time-scale modifications. The analysis technique is based on narrow band filtering using special analysis filters with frequency-modulated impulse response. The main advantage of the technique is high accuracy of harmonic parameters estimation that provides adequate harmonic/noise separation and artifact free implementing of voice modifications. The processing methods described in the paper are based on the harmonic+noise model.
Download Automated Equalization for Room Resonance Suppression
Estimating room resonances in locations of big events and looking for counter-measures are normally done by sound engineers, mainly before the beginning of the event. In this paper an automation to enhance the audio quality in event rooms by suppressing the room resonances with a parametric equalizer of several high-Q peak filters is proposed. The room characteristics can be identified with few measurements in the listening area during the event, without applying an additional measuring signal (using its original sound signal). Based on this room characteristics the equalization filters are automatically designed. The results of several rooms tested with the automated equalization for room resonance suppression are presented as well as a discussion on the covered topics.
Download The Influence of Small Variations in a Simplified Guitar Amplifier Model
A strongly simplified guitar amplifier model, consisting of four stages, is presented. The exponential sweep technique is used to measure the frequency dependent harmonic spectra. The influence of small variations of the system parameters on the harmonic components is analyzed. The differences of the spectra are explained and visualized.
Download Informed Selection of Frames for Music Similarity Computation
In this paper we present a new method to compute frame based audio similarities, based on nearest neighbour density estimation. We do not recommend it is as a practical method for large collections because of the high runtime. Rather, we use this new method for a detailed analysis to get a deeper insight on how a bag of frames approach (BOF) determines similarities among songs, and in particular, to identify those audio frames that make two songs similar from a machine’s point of view. Our analysis reveals that audio frames of very low energy, which are of course not the most salient with respect to human perception, have a surprisingly big influence on current similarity measures. Based on this observation we propose to remove these low-energy frames before computing song models and show, via classification experiments, that the proposed frame selection strategy improves the audio similarity measure.
Download Improvement of Acoustic Localization Using the STSA denoising with a novel Suppression Rule
This paper proposes innovative de-noise filters in a framework, whose aim is the localization of an acoustic source in a noisy environment. The main focuses are the automatic detection of transient sound events and the separation of the events of interest from the noise. A microphone array is used to capture timespatial information and an adaptive filter can be initialized to learn the ambient noise spectrum when signals of interest are absent. We propose an algorithm based on the Short Time Spectral Attenuation method to remove the noise from each sensor of the array, before the source localization task is performed. The Time Difference Of Arrival (TDOA) methods are used for multiple sources localization. The experimental results show the efficiency of our framework in stationary noisy environments.
Download Re-targeting Expressive Musical Style Using a Machine-Learning Method
Expressive musical performing style involves more than what is simply represented on the score. Performers imprint their personal style on each performances based on their musical understanding. Expressive musical performing style makes the music come alive by shaping the music through continuous variation. It is observed that the musical style can be represented by appropriate numerical parameters, where most parameters are related to the dynamics. It is also observed that performers tends to perform music sections and motives of similar shape in similar ways, where music sections and motives can be identified by an automatic phrasing algorithm. An experiment is proposed for producing expressive music from raw quantized music files using machine-learning methods like Support Vector Machines. Experimental results show that it is possible to induce some of a performer’s style by using the music parameters extracted from the audio recordings of their real performance.