Download Separating Piano Recordings into Note Events Using a Parametric Imitation Approach In this paper we present a working system for separating a piano recording into events representing individual piano notes. Each note is parameterized with a transient-plus-harmonics model that, should all the parameters be reliably estimated, would produce near perfect reconstruction for each note as well as for the whole recording. However, interference between overlapping notes makes it hard to estimate parameters from their combination. In this work we propose to assess the estimability of sinusoidal parameters via their apparent degree of interference, estimate the estimable ones using algorithms suitable for different interference situations, and infer the hard-to-estimate parameters from the estimated ones. The outcome is a sequence of separate, parameterized piano notes that perceptually highly resemble, if are not identical to, the notes in the original recording. This allows for later analysis and processing stages using algorithms designed for separate notes.
Download Simulation of Analog Flanger Effect Using BBD Circuit This paper deals with simulation of BBD circuit based analog flanger effects. The famous Electro-Harmonix Deluxe Electric Mistress flanger effect was used as a case study in this paper. The main attention of this paper is paid to the analysis and simulation of the LFO circuit, the BBD clock generator circuit and BBD circuit simulation of this effect. However, in order to compare the simulation results with measured data, the signal path simulation using the DK-method has been introduced as well.
Download Automatic Violin Synthesis Using Expressive Musical Term Features The control of interpretational properties such as duration, vibrato, and dynamics is important in music performance. Musicians continuously manipulate such properties to achieve different expressive intentions. This paper presents a synthesis system that automatically converts a mechanical, deadpan interpretation to distinct expressions by controlling these expressive factors. Extending from a prior work on expressive musical term analysis, we derive a subset of essential features as the control parameters, such as the relative time position of the energy peak in a note and the mean temporal length of the notes. An algorithm is proposed to manipulate the energy contour (i.e. for dynamics) of a note. The intended expressions of the synthesized sounds are evaluated in terms of the ability of the machine model developed in the prior work. Ten musical expressions such as Risoluto and Maestoso are considered, and the evaluation is done using held-out music pieces. Our evaluations show that it is easier for the machine to recognize the expressions of the synthetic version, comparing to those of the real recordings of an amateur student. While a listening test is under construction as a next step for further performance validation, this work represents to our best knowledge a first attempt to build and quantitatively evaluate a system for EMT analysis/synthesis.
Download Synthesis of Sound Textures with Tonal Components Using Summary Statistics and All-Pole Residual Modeling The synthesis of sound textures, such as flowing water, crackling fire, an applauding crowd, is impeded by the lack of a quantitative definition. McDermott and Simoncelli proposed a perceptual source-filter model using summary statistics to create compelling synthesis results for non-tonal sound textures. However, the proposed method does not work well with tonal components. Comparing the residuals of tonal sound textures and non-tonal sound textures, we show the importance of residual modeling. We then propose a method using auto regressive modeling to reduce the amount of data needed for resynthesis and delineate a modified method for analyzing and synthesizing both tonal and non-tonal sound textures. Through user evaluation, we find that modeling the residuals increases the realism of tonal sound textures. The results suggest that the spectral content of the residuals has an important role in sound texture synthesis, filling the gap between filtered noise and sound textures as defined by McDermott and Simoncelli. Our proposed method opens possibilities of applying sound texture analysis to musical sounds such as rapidly bowed violins.
Download Monophonic Pitch Detection by Evaluation of Individually Parameterized Phase Locked Loops This paper describes a new efficient and sample based monophonic pitch tracking approach using multiple phase locked loops (PLLs). Hereby, distinct subband signals traverse pairs of individually parameterized PLLs. Based on the relation of the instantaneous pitch sample of respective PLLs to one another, relevant features per pitch candidate are derived. These features are combined into pitch candidate scores. Pitch candidates which exhibit the maximum score per sampling instance and exceed a voicing threshold, contribute to the overall pitch track. Evaluations with up to date datasets show that the tracking performance, compared to implementations which use only one PLL has significantly improved and nearly approaches the scores of a state of the art monophonic pitch tracker.
Download Reducing the Aliasing of Nonlinear Waveshaping Using Continuous-Time Convolution Nonlinear waveshaping is a common technique in musical signal processing, both in a static memoryless context and within feedback systems. Such waveshaping is usually applied directly to a sampled signal, generating harmonics that exceed the Nyquist frequency and cause aliasing distortion. This problem is traditionally tackled by oversampling the system. In this paper, we present a novel method for reducing this aliasing by constructing a continuous-time approximation of the discrete-time signal, applying the nonlinearity to it, and filtering in continuous-time using analytically applied convolution. The presented technique markedly reduces aliasing distortion, especially in combination with low order oversampling. The approach is also extended to allow it to be used within a feedback system.
Download Real-Time Audio Visualization With Reassigned Non-uniform Filter Banks Filter banks, both uniform and non-uniform, are widely used for signal analysis and processing. However, the application of a timefrequency localized filter inevitably causes some amount of spectral and temporal leakage that, simultaneously, cannot be arbitrarily reduced. Reassignment is a classical procedure to eliminate this leakage in short-time Fourier spectrograms, thereby providing a sharper, more exact time-frequency domain signal representation. The reassignment technique was recently generalized to general filter banks, opening new possibilities for its application in signal analysis and processing. We present here the very first implementation of filter bank reassignment in a real-time analysis setting, more specifically as visualization in a basic audio player application. The visualization provides a low delay moving spectrogram with respect to virtually any time-frequency filter bank by interfacing the C backend of the LTFAT open-source toolbox for time-frequency processing. Low delay is achieved by blockwise processing, implemented with the JUCE C++ Library.