Download Onset Detection Revisited Various methods have been proposed for detecting the onset times of musical notes in audio signals. We examine recent work on onset detection using spectral features such as the magnitude, phase and complex domain representations, and propose improvements to these methods: a weighted phase deviation function and a halfwave rectified complex difference. These new algorithms are compared with several state-of-the-art algorithms from the literature, and these are tested using a standard data set of short excerpts from a range of instruments (1060 onsets), plus a much larger data set of piano music (106054 onsets). Some of the results contradict previously published results and suggest that a similarly high level of performance can be obtained with a magnitude-based (spectral flux), a phase-based (weighted phase deviation) or a complex domain (complex difference) onset detection function.
Download Design principles for lumped model discretisation using Möbius transforms Computational modelling of audio systems commonly involves discretising lumped models. The properties of common discretisation schemes are typically derived through analysis of how the imaginary axis on the Laplace-transform s-plane maps onto the Ztransform z-plane and the implied stability regions. This analysis ignores some important considerations regarding the mapping of individual poles, in particular the case of highly-damped poles. In this paper, we analyse the properties of an extended class of discretisations based on Möbius transforms, both as mappings and discretisation schemes. We analyse and extend the concept of frequency warping, well-known in the context of the bilinear transform, and we characterise the relationship between the damping and frequencies of poles in the s- and z-planes. We present and analyse several design criteria (damping monotonicity, stability) corresponding to desirable properties of the discretised system. Satisfying these criteria involves selecting appropriate transforms based on the pole structure of the system on the s-plane. These theoretical developments are finally illustrated on a diode clipper nonlinear model.
Download Time-Variant Gray-Box Modeling of a Phaser Pedal A method to measure the response of a linear time-variant (LTV) audio system is presented. The proposed method uses a series of short chirps generated as the impulse response of several cascaded allpass filters. This test signal can measure the characteristics of an LTV system as a function of time. Results obtained from testing of this method on a guitar phaser pedal are presented. A proof of concept gray-box model of the measured system is produced based on partial knowledge about the internal structure of the pedal and on the spectral analysis of the measured responses. The temporal behavior of the digital model is shown to be very similar to that of the measured device. This demonstrates that it is possible to measure LTV analog audio systems and produce approximate virtual analog models based on these results.
Download Exact Discrete-Time Realization of a Dolby B Encoding/Decoding Architecture An algebraic technique which computes nonlinear, delay-free digital filter networks is applied to model the Dolby B in the discretetime. The model preserves the topology of the analog system, and imports the characteristics of the nonlinear processing blocks which are responsible of the peculiar functioning of Dolby B. The resulting numerical system exhibits qualitatively similar dynamic behavior and performance – full compliance with the Dolby B specifications would be achieved by deriving, from comprehensive data sheets of the system, accurate discrete-time models of the analog processing blocks. Results demonstrate that the computation converges if proper iterative methods are employed.
Download Recent developments in PWSYNTH PWSynth was originally a visual synthesis language situated in PatchWork. Recently our research team has started a complete rewrite of the system so that it can be adapted to our new programming environment called PWGL. In this paper we present the main differences of the old and new systems. These include switching from C to C++, efficiency issues, interface between PWGL and the synthesis engine, and a novel copy-synth-patch scheme.
Download Extracting automatically the perceived intensity of music titles We address the issue of extracting automatically high-level musical descriptors out of their raw audio signal. This work focuses on the extraction of the perceived intensity of music titles, that evaluates how energic the music is perceived by listeners. We present here first the perceptive tests that we have conducted, in order to evaluate the relevance and the universality of the perceived intensity descriptor. Then we present several methods used to extract relevant features used to build automatic intensity extractors: usual Mpeg7 low level features, empirical method, and features automatically found using our Extractor Discovery System (EDS), and compare the final performances of their extractors.
Download Sound Effects for a Silent Computer System This paper proposes the sonification of the activity of a computer system that allows the user to monitor the basic performance parameters of the system, like CPU load, read and write activity of the hard disk or network traffic. Although, current computer systems still produce acoustic background noise, future and emerging computer systems will be more and more optimized with respect to their noise emission. In contrast to most of the concepts of auditory feedback, which present a particular sound as a feedback to a user’s command, the proposed feedback is mediated by the running computer system. The user’s interaction stimulates the system and hence the resulting feedback offers more realistic information about the current states of performance of the system. On the one hand the proposed sonification can mimic the acoustical behavior of operating components inside a computer system, while on the other hand, new qualities can be synthesized that enrich interaction with the device. Different forms of sound effects and generation for the proposed auditory feedback are realized to experiment with the usage in an environment of silent computer systems.
Download Circle Maps as a Simple Oscillators for Complex Behavior: II. Experiments The circle map is a general non-linear iterated function that maps the circle onto itself. In its standard form it can be interpreted as a simple sinusoidal oscillator which is perturbed by a non-linear term. By varying the strength of the non-linear contribution a rich array of non-linear responses can be achieved, including waveshaping, pitch-bending, period-doubling and highly irregular patterns. We describe a number of such examples and discuss their subjective auditory perception.
Download Simulation of the Diode Limiter in Guitar Distortion Circuits by Numerical Solution of Ordinary Differential Equations The diode clipper circuit with an embedded low-pass filter lies at the heart of both diode clipping “Distortion” and “Overdrive” or “Tube Screamer” effects pedals. An accurate simulation of this circuit requires the solution of a nonlinear ordinary differential equation (ODE). Numerical methods with stiff stability – Backward Euler, Trapezoidal Rule, and second-order Backward Difference Formula – allow the use of relatively low sampling rates at the cost of accuracy and aliasing. However, these methods require iteration at each time step to solve a nonlinear equation, and the tradeoff for this complexity must be evaluated against simple explicit methods such as Forward Euler and fourth order Runge-Kutta, which require very high sampling rates for stability. This paper surveys and compares the basic ODE solvers as they apply to simulating circuits for audio processing. These methods are compared to a static nonlinearity with a pre-filter. It is found that implicit or semiimplicit solvers are preferred and that the filter/static nonlinearity approximation is often perceptually adequate.
Download Real-time Auralisation System for Virtual Microphone Positioning A computer application was developed to simulate the process of microphone positioning in sound recording applications. A dense, regular grid of impulse responses pre-recorded on the region of the room under study allowed the sound captured by a virtual microphone to be auralised through real-time convolution with an anechoic stream representing the sound source. Convolution was performed using a block-based variation on the overlap-add method where the summation of many small subconvolutions produced each block of output data samples. As the applied RIR filter varied on successive audio output blocks, a short cross fade was applied to avoid glitches in the audio. The maximum possible length of impulse response applied was governed by the size of audio processing block (hence latency) employed by the program. Larger blocks allowed a lower processing time per sample. At 23.2ms latency (1024 samples at 44.1kHz), it was possible to apply 9 second impulse responses on a standard laptop computer.