Download Musical Signal Analysis Using Fractional-Delay Inverse Comb Filters A novel filter configuration for the analysis of harmonic musical signals is proposed. The method is based on inverse comb filtering that allows for the extraction of selected harmonic components or the background noise component between the harmonic spectral components. A highly accurate delay required in the inverse comb filter is implemented with a high-order allpass filter. The paper shows that the filter is easy to design, efficient to implement, and it enables accurate low-level feature analysis of musical tones. We describe several case studies to demonstrate the effectiveness of the proposed approach: isolating a single partial from a synthetic signal, analyzing the even-to-odd ratio of harmonics in a clarinet tone, and extracting the residual from a bowed string tone.
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 Alias-free Virtual Analog Oscillators Using a Feedback Delay Loop The rich spectra of classic waveforms (sawtooth, square and triangular) are obtained by discontinuities in the waveforms or their derivatives. At the same time, the discontinuities lead to aliasing when the waveforms are digitally generated. To remove or reduce the aliasing, researchers have proposed various methods, mostly based on limiting bandwidth or smoothing the waveforms. This paper introduces a new approach to generate the virtual analog oscillators with no aliasing. The approach relies on generating an impulse train using a feedback delay loop, often used for the physical modeling of musical instruments. Classic waveforms are then derived from the impulse train with a leaky integrator. Although the output generated by this method is not exactly periodic, it perceptually sounds harmonic. While additional processing is required for time-varying pitch shifting, resulting in some high-frequency attenuation when the pitch changes, the proposed method is computationally more efficient than other algorithms and the high-frequency attenuation can be also adjusted.
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 Simulating Idiomatic Playing Styles in a Classical Guitar Synthesizer: Rasgueado as a Case Study This paper presents our research efforts to synthesize complex instrumental gestures using a score-based control scheme. Our specific goal is to simulate the rasgueado technique that is popular especially in flamenco music. This technique is also used in the classical guitar repertoire. Rasgueado is especially challenging as ordinary music notation is not adequate to represent the dense stream of notes required for a convincing simulation. We will take two approaches to realize our task. First, we use the practical knowledge of how the actual performance is accomplished by the human player. A second, complementary, approach is to analyze an excerpt from real guitar playing. Our main focus here is to extract the onset times and the amplitudes of the recoded gesture. Next we combine the results from the two analysis steps using a constraintbased approach to find possible pitch and fingering sequences. Finally we translate the findings to our macro-note scheme that allows us to fill algorithmically a musical score.
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 Modeling of the Carbon Microphone Nonlinearity for a Vintage Telephone Sound Effect The telephone sound effect is widely used in music, television and the film industry. This paper presents a digital model of the carbon microphone nonlinearity which can be used to produce a vintage telephone sound effect. The model is constructed based on measurements taken from a real carbon microphone. The proposed model is a modified version of the sandwich model previously used for nonlinear telephone handset modeling. Each distortion component can be modeled individually based on the desired features. The computational efficiency can be increased by lumping the spectral processing of the individual distortion components together. The model incorporates a filtered noise source to model the self-induced noise generated by the carbon microphones. The model has also an input level depended noise generator for additional sound quality degradation. The proposed model can be used in various ways in the digital modeling of the vintage telephone sound.
Download Automated Calibration of a Parametric Spring Reverb Model The calibration of a digital spring reverberator model is crucial for the authenticity and quality of the sound produced by the model. In this paper, an automated calibration of the model parameters is proposed, by analysing the spectrogram, the energy decay curve, the spectrum, and the autocorrelation of the time signal and spectrogram. A visual inspection of the spectrograms as well as a comparison of sound samples proves the approach to be successful for estimating the parameters of reverberators with one, two and three springs. This indicates that the proposed method is a viable alternative to manual calibration of spring reverberator models.
Download Vector Phaseshaping Synthesis This paper introduces the Vector Phaseshaping (VPS) synthesis technique, which extends the classic Phase Distortion method by providing flexible means to distort the phase of a sinusoidal oscillator. This is achieved by describing the phase distortion function using one or more breakpoint vectors, which are then manipulated in two dimensions to produce waveshape modulation at control and audio rates. The synthesis parameters and their effects are explained, and the spectral description of the method is derived. Certain synthesis parameter combinations result in audible aliasing, which can be reduced with a novel aliasing suppression algorithm described in the paper. The extension is capable of producing a variety of interesting harmonic and inharmonic spectra, including for instance, formant peaks, while the two-dimensional form of the control parameters is expressive and is well suited for interactive applications.