9,941 research outputs found

    Computer Models for Musical Instrument Identification

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    PhDA particular aspect in the perception of sound is concerned with what is commonly termed as texture or timbre. From a perceptual perspective, timbre is what allows us to distinguish sounds that have similar pitch and loudness. Indeed most people are able to discern a piano tone from a violin tone or able to distinguish different voices or singers. This thesis deals with timbre modelling. Specifically, the formant theory of timbre is the main theme throughout. This theory states that acoustic musical instrument sounds can be characterised by their formant structures. Following this principle, the central point of our approach is to propose a computer implementation for building musical instrument identification and classification systems. Although the main thrust of this thesis is to propose a coherent and unified approach to the musical instrument identification problem, it is oriented towards the development of algorithms that can be used in Music Information Retrieval (MIR) frameworks. Drawing on research in speech processing, a complete supervised system taking into account both physical and perceptual aspects of timbre is described. The approach is composed of three distinct processing layers. Parametric models that allow us to represent signals through mid-level physical and perceptual representations are considered. Next, the use of the Line Spectrum Frequencies as spectral envelope and formant descriptors is emphasised. Finally, the use of generative and discriminative techniques for building instrument and database models is investigated. Our system is evaluated under realistic recording conditions using databases of isolated notes and melodic phrases

    Waveguide physical modeling of vocal tract acoustics: flexible formant bandwidth control from increased model dimensionality

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    Digital waveguide physical modeling is often used as an efficient representation of acoustical resonators such as the human vocal tract. Building on the basic one-dimensional (1-D) Kelly-Lochbaum tract model, various speech synthesis techniques demonstrate improvements to the wave scattering mechanisms in order to better approximate wave propagation in the complex vocal system. Some of these techniques are discussed in this paper, with particular reference to an alternative approach in the form of a two-dimensional waveguide mesh model. Emphasis is placed on its ability to produce vowel spectra similar to that which would be present in natural speech, and how it improves upon the 1-D model. Tract area function is accommodated as model width, rather than translated into acoustic impedance, and as such offers extra control as an additional bounding limit to the model. Results show that the two-dimensional (2-D) model introduces approximately linear control over formant bandwidths leading to attainable realistic values across a range of vowels. Similarly, the 2-D model allows for application of theoretical reflection values within the tract, which when applied to the 1-D model result in small formant bandwidths, and, hence, unnatural sounding synthesized vowels

    Extended pipeline for content-based feature engineering in music genre recognition

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    We present a feature engineering pipeline for the construction of musical signal characteristics, to be used for the design of a supervised model for musical genre identification. The key idea is to extend the traditional two-step process of extraction and classification with additive stand-alone phases which are no longer organized in a waterfall scheme. The whole system is realized by traversing backtrack arrows and cycles between various stages. In order to give a compact and effective representation of the features, the standard early temporal integration is combined with other selection and extraction phases: on the one hand, the selection of the most meaningful characteristics based on information gain, and on the other hand, the inclusion of the nonlinear correlation between this subset of features, determined by an autoencoder. The results of the experiments conducted on GTZAN dataset reveal a noticeable contribution of this methodology towards the model's performance in classification task.Comment: ICASSP 201

    Classification of Musical Instruments sounds by Using MFCC and Timbral Audio Descriptors

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    Identification of the musical instrument from a music piece is becoming area of interest for researchers in recent years. The system for identification of musical instrument from monophonic audio recording is basically performs three tasks: i) Pre-processing of inputted music signal; ii) Feature extraction from the music signal; iii) Classification. There are many methods to extract the audio features from an audio recording like Mel-frequency Cepstral Coefficients (MFCC), Linear Predictive Codes (LPC), Linear Predictive Cepstral Coefficients (LPCC), Perceptual Linear Predictive Coefficients (PLP), etc. The paper presents an idea to identify musical instruments from monophonic audio recordings by extracting MFCC features and timbre related audio descriptors. Further, three classifiers K-Nearest Neighbors (K-NN), Support Vector Machine (SVM) and Binary Tree Classifier (BT) are used to identify the musical instrument name by using feature vector generated in feature extraction process. The analysis is made by studying results obtained by all possible combinations of feature extraction methods and classifiers. Percentage accuracies for each combination are calculated to find out which combinations can give better musical instrument identification results. The system gives higher percentage accuracies of 90.00%, 77.00% and 75.33% for five, ten and fifteen musical instruments respectively if MFCC is used with K-NN classifier and for Timbral ADs higher percentage accuracies of 88.00%, 84.00% and 73.33% are obtained for five, ten and fifteen musical instruments respectively if BT classifier is used. DOI: 10.17762/ijritcc2321-8169.150713

    Sequential Complexity as a Descriptor for Musical Similarity

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    We propose string compressibility as a descriptor of temporal structure in audio, for the purpose of determining musical similarity. Our descriptors are based on computing track-wise compression rates of quantised audio features, using multiple temporal resolutions and quantisation granularities. To verify that our descriptors capture musically relevant information, we incorporate our descriptors into similarity rating prediction and song year prediction tasks. We base our evaluation on a dataset of 15500 track excerpts of Western popular music, for which we obtain 7800 web-sourced pairwise similarity ratings. To assess the agreement among similarity ratings, we perform an evaluation under controlled conditions, obtaining a rank correlation of 0.33 between intersected sets of ratings. Combined with bag-of-features descriptors, we obtain performance gains of 31.1% and 10.9% for similarity rating prediction and song year prediction. For both tasks, analysis of selected descriptors reveals that representing features at multiple time scales benefits prediction accuracy.Comment: 13 pages, 9 figures, 8 tables. Accepted versio
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