Download Identification of individual guitar sounds by support vector machines This paper introduces an automatic classification system for the identification of individual classical guitars by single notes played on these guitars. The classification is performed by Support Vector Machines (SVM) that have been trained with the features of the single notes. The features used for classification were the time series of the partial tones, the time series of the MFCCs (Mel Frequency Cepstral Coefficients), and the “nontonal” contributions to the spectrum. The influences of these features on the classification success are reported. With this system, 80% of the sounds recorded with three different guitars were classified correctly. A supplementary classification experiment was carried out with human listeners resulting in a rate of 65% of correct classifications.
Download Inferring the hand configuration from hand clapping sounds In this paper, a technique for inferring the configuration of a clapper’s hands from a hand clapping sound is described. The method was developed based on analysis of synthetic and recorded hand clap sounds, labeled with the corresponding hand configurations. A naïve Bayes classifier was constructed to automatically classify the data using two different feature sets. The results indicate that the approach is applicable for inferring the hand configuration.
Download Analysis of piano tones using an inharmonic inverse comb filter This paper presents a filter configuration for canceling and separating partials from inharmonic piano tones. The proposed configuration is based on inverse comb filtering, in which the delay line is replaced with a high-order filter that has a proper phase response. Two filter design techniques are tested with the method: an FIR filter, which is designed using frequency sampling, and an IIR filter, which consists of a set of second-order allpass filters that match the desired group delay. It is concluded that it is possible to obtain more accurate results with the FIR filter, while the IIR filter is computationally more efficient. The paper shows that the proposed analysis method provides an effective and easy way of extracting the residual signal and selecting partials from piano tones. This method is suitable for analysis of recorded piano tones.
Download Detecting arrivals within room impulse responses using matching pursuit This paper proposes to use Matching Pursuit, in order to investigate some statistical foundations of Room Acoustics, such as the temporal distribution of arrivals, and the estimation of mixing time. As this has never been experimentally explored, this study is a first step towards a validation of the ergodic theory of reverberation. The use of Matching Pursuit is implicit, since correlation between the impulse response and the direct sound is assumed. The compensation for the energy decay is necessary to obtain stationnary signals. Methods for determining the best the temporal boundaries of the direct sound, for choosing an appropriate stopping criteria based on the similarity between acoustical indices of the original RIR and those of the synthesized signal, and for experimentally defining the mixing time constitute the scope of this study.
Download On the control of the phase of resonant filters with applications to percussive sound modeling Source-filter models are widely used in numerous audio processing fields, from speech processing to percussive/contact sound synthesis. The design of filters for these models—be it from scratch or from spectral analysis—usually involves tuning frequency and damping parameters and/or providing an all-pole model of the resonant part of the filter. In this context, and for the modelling of percussive (non-sustained) sounds, a source signal can be estimated from a filtered sound through a time-domain deconvolution process. The result can be plagued with artifacts when resonances exhibit very low bandwidth and lie very close in frequency. We propose in this paper a method that noticeably reduces the artifacts of the deconvolution process through an inter-resonance phase synchronization. Results show that the proposed method is able to design filters inducing fewer artifacts at the expense of a higher dynamic range.
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 Score level timbre transformations of violin sounds The ability of a sound synthesizer to provide realistic sounds depends to a great extent on the availability of expressive controls. One of the most important expressive features a user of the synthesizer would desire to have control of, is timbre. Timbre is a complex concept related to many musical indications in a score such as dynamics, accents, hand position, string played, or even indications referring timbre itself. Musical indications are in turn related to low level performance controls such as bow velocity or bow force. With the help of a data acquisition system able to record sound synchronized to performance controls and aligned to the performed score and by means of statistical analysis, we are able to model the interrelations among sound (timbre), controls and musical score indications. In this paper we present a procedure for score-controlled timbre transformations of violin sounds within a sample based synthesizer. Given a sound sample and its trajectory of performance controls: 1) a transformation of the controls trajectory is carried out according to the score indications, 2) a new timbre corresponding to the transformed trajectory is predicted by means of a timbre model that relates timbre with performance controls and 3) the timbre of the original sound is transformed by applying a timevarying filter calculated frame by frame as the difference of the original and predicted envelopes.
Download Audio analysis in PWGLSynth In this paper, we present an incremental improvement of a known fundamental frequency estimation algorithm for monophonic signals. This is viewed as a case study of using our signal graph based synthesis language, PWGLSynth, for audio analysis. The roles of audio and control signals are discussed in both analysis and synthesis contexts. The suitability of the PWGLSynth system for this field of applications is examined and some problems and future work is identified.
Download Vocal melody detection in the presence of pitched accompaniment using harmonic matching methods Vocal music is characterized by a melodically salient singing voice accompanied by one or more instruments. With a pitched instrument background, multiple periodicities are simultaneously present and the task becomes one of identifying and tracking the vocal pitch based on pitch strength and smoothness constraints. Frequency domain harmonic matching methods can be applied to detect pitch via the harmonically related frequencies that fit the signal’s measured spectral peaks. The specific spectral fitness measure is expected to influence the performance of vocal pitch detection depending on the nature of the polyphonic mixture. In this work, we consider Indian classical music which provides important examples of singing voice accompanied by strongly pitched instruments. It is shown that the spectral fitness measure of the two-way mismatch method is well suited to track vocal pitch in the presence of the pitched percussion with its strong but sparse harmonic structure. The detected pitch is further used to obtain a measure of voicing that reliably discriminates vocal segments from purely instrumental regions.
Download Inverting dynamics compression with minimal side information Dynamics processing is a widespread technique, both at music production and diffusion stages. In particular, dynamic compression is often used in such a way that the “average” listener can best enjoy the music. However, this may lead to an excessive use of compression, especially with respect to listeners in quiet listening conditions. This paper presents estimates on the amount of extra data that is needed to invert the effects of such non-linear processing, using simple blind identification techniques. We present two simple test cases, first in the case when perfect reconstruction is needed, and second when the ancillary data rate is constrained, leading to an approximate reconstruction.