Download Coefficient-modulated first-order allpass filter as distortion effect
A novel approach to implement the distortion effect is introduced. The proposed approach is based on time-varying phase distortion of the input signal, and it is implemented using a coefficient modulated first-order allpass filter. This new technique provides control over the distorted band with a proper choice of the modulating signal. By choosing a modulating signal that applies the phase distortion only for low frequencies, the aliasing often generated by conventional distortion effects, which modify the signal amplitude, can be greatly reduced. Modulation signals that produce distortion effects applicable for electric guitar playing are also discussed. Sound examples on the use of the filter can be found at http://www.acoustics.hut.fi/~jpekonen/ Papers/dafx08/.
Download Adaptive Phase Distortion Synthesis
This article discusses Phase Distortion synthesis and its application to arbitrary input signals. The main elements that compose the technique are presented. Its similarities to Phase Modulation are discussed and the equivalence between the two techniques is explored. Two alternative methods of distorting the phase of an arbitrary signal are presented. The first is based on the audio-rate modulation of a first-order allpass filter coefficient. The other method relies on a re-casting of the Phase Modulation equation, which leads to a heterodyned form of waveshaping. The relationship of these implementations to the original technique is explored in detail. Complementing the article, a number of examples are discussed, demonstrating the application of the technique as an interesting digital audio effect.
Download Sound synthesis using an allpass filter chain with audio‐rate coefficient modulation
This paper describes a sound synthesis technique that modulates the coefficients of allpass filter chains using audio-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM–like spectrum, and that its bandwidth is extended and further processed by non-sinusoidal FM when the sections are cascaded. The cascade length parameter provides dynamic bandwidth control to prevent upper range aliasing artifacts, and the amount of spectral content within that band can be controlled using a modulation index parameter. The technique is capable of synthesizing rich and evolving timbres, including those resembling classic virtual analog waveforms. It can also be used as an audio effect with pitch-tracked input sources. Software and sound examples are available at http://www.acoustics.hut.fi/publications/papers/dafx09-cm/
Download Spectral Dealy Filters with Feedback Delay Filters with Feedback and Time-Varying Coefficients
A recently introduced structure to implement a continuously smooth spectral delay, based on a cascade of first-order allpass filters and an equalizing filter, is described and the properties of this spectral delay filter are reviewed. A new amplitude envelope equalizing filter for the spectral delay filter is proposed and the properties of structures utilizing feedback and/or time-varying filter coefficients are discussed. In addition, the stability conditions for the feedback and the time-varying structures are derived. A spectral delay filter can be used for synthesizing chirp-like sounds or for modifying the timbre of arbitrary audio signals. Sound examples on the use of the spectral delay filters utilizing the structures discussed in this paper can be found at http://www.acoustics.hut. fi/publications/papers/dafx09-sdf/.
Download On Minimizing the Look-Up Table Size in Quasi-Bandlimited Classical Waveform Oscillators
In quasi-bandlimited classical waveform oscillators, the aliasing distortion present in a trivially sampled waveform can be reduced in the digital domain by applying a tabulated correction function. This paper presents an approach that applies the correction function in the differentiated domain by synthesizing a bandlimited impulse train (BLIT) that is integrated to obtain the desired bandlimited waveform. The ideal correction function of the BLIT method is infinitely long and in practice needs to be windowed. In order to obtain a good alias-reduction performance, long tables are typically required. It is shown that when a short look-up table is used, a windowed ideal correction function does not provide the best performance in terms of minimizing aliasing audibility. Instead, audibly improved alias-reduction performance can be obtained using a look-up table that has a parametric control over the low-order generations of aliasing. Some practical parametric look-up table designs are discussed in this paper, and their use and alias-reduction performance are exemplified. The look-up table designs discussed in this paper providing the best alias-reduction performance are parametric window functions and least-squares optimized multiband FIR filter designs.
Download Digital Emulation of Distortion Effects by Wave and Phase Shaping Methods
This paper will consider wave (amplitude) and phase signal shaping techniques for the digital emulation of distortion effect processing. We examine how to determine the Wave- and Phaseshaping functions with harmonic amplitude and phase data. Three distortion effects units are used to provide test data. The action of the Wave- and Phase- shaping functions derived for these effects is demonstrated with the assistance of a superresolution frequency-domain analysis technique.
Download Computationally Efficient Hammond Organ Synthesis
The Hammond organ is an early electronic musical instrument, which was popular in the 1960s and 1970s. This paper proposes computationally efficient models for the Hammond organ and its rotating speaker system, the Leslie. Organ tones are generated using additive synthesis with appropriate features, such as a typical fast attack and decay envelope for the weighted sum of the harmonics and a small amplitude modulation simulating the construction inaccuracies of tone wheels. The key click is realized by adding the sixth harmonic modulated by an additional envelope to the original organ tone. For the Leslie speaker modeling we propose a new approach, which is based on time-varying spectral delay filters producing the Doppler effect. The resulting virtual organ, which is conceptually easy, has a pleasing sound and is computationally efficient to implement.
Download Virtual Analog Oscillator Hard Synchronisation: Fourier series and an efficient implementation
This paper investigates a number of digital methods to produce the Analog subtractive synthesis effect of ‘Hard Synchronisation.’ While the original effect is produced by an explicit waveform phase reset, other approaches are given that produce an equivalent output. In particular, based on measurements taken from a real-analog synthesizer, a comb filtering model is proposed. This description ties in with earlier work but here an explicit structure is provided. This filter-based approach is then shown to be far more computationally efficient than the synchronisation by phase reset. This efficiency is at a minor cost as it is shown that it has a minimal impact on the sonic accuracy.
Download Nonlinear-Phase Basis Functions in Quasi-Bandlimited Oscillator Algorithms
Virtual analog synthesis requires bandlimited source signal algorithms. An efficient methodology for the task expresses the traditionally used source waveforms or their time-derivatives as a sequence of bandlimited impulses or step functions. Approximations of the ideal bandlimited functions used in these quasi-bandlimited oscillator algorithms are typically linear-phase functions. In this paper, a general nonlinear-phase approach to the task is proposed. The discussed technique transforms an analog prototype filter to a digital filter using a modified impulse invariance transformation method that enables the impulse response to be sampled with arbitrary sub-sample shifts. The resulting digital filter is a set of parallel first- and/or second-order filters that are excited with short burst-like signals that depend on the offset of the waveform discontinuities. The discussed approach is exemplified with a number of design cases, illustrating different trade-offs between good alias reduction and low computational cost.