3,128 research outputs found
Musical instrument classification using non-negative matrix factorization algorithms
In this paper, a class of algorithms for automatic classification of individual musical instrument sounds is presented. Several perceptual features used in general sound classification applications were measured for 300 sound recordings consisting of 6 different musical instrument classes (piano, violin, cello, flute, bassoon and soprano saxophone). In addition, MPEG-7 basic spectral and spectral basis descriptors were considered, providing an effective combination for accurately describing the spectral and timbrai audio characteristics. The audio flies were split using 70% of the available data for training and the remaining 30% for testing. A classifier was developed based on non-negative matrix factorization (NMF) techniques, thus introducing a novel application of NMF. The standard NMF method was examined, as well as its modifications: the local, the sparse, and the discriminant NMF. Experimental results are presented to compare MPEG-7 spectral basis representations with MPEG-7 basic spectral features alongside the various NMF algorithms. The results indicate that the use of the spectrum projection coefficients for feature extraction and the standard NMF classifier yields an accuracy exceeding 95%. Ā©2006 IEEE
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Musical instrument classification using non-negative matrix factorization algorithms and subset feature selection
In this paper, a class of algorithms for automatic classification of individual musical instrument sounds is presented. Several perceptual features used in sound classification applications as well as MPEG-7 descriptors were measured for 300 sound recordings consisting of 6 different musical instrument classes. Subsets of the feature set are selected using branchand-bound search, obtaining the most suitable features for classification. A class of classifiers is developed based on the non-negative matrix factorization (NMF). The standard NMF method is examined as well as its modifications: the local, the sparse, and the discriminant NMF. The experimental results compare feature subsets of varying sizes alongside the various NMF algorithms. It has been found that a subset containing the mean and the variance of the first mel-frequency cepstral coefficient and the AudioSpectrumFlatness descriptor along with the means of the AudioSpectrumEnvelope and the AudioSpectrumSpread descriptors when is fed to a standard NMF classifier yields an accuracy exceeding 95%
Wavelet-based techniques for speech recognition
In this thesis, new wavelet-based techniques have been developed for the
extraction of features from speech signals for the purpose of automatic speech
recognition (ASR). One of the advantages of the wavelet transform over the short
time Fourier transform (STFT) is its capability to process non-stationary signals.
Since speech signals are not strictly stationary the wavelet transform is a better
choice for time-frequency transformation of these signals. In addition it has
compactly supported basis functions, thereby reducing the amount of
computation as opposed to STFT where an overlapping window is needed. [Continues.
Audio Signal Processing Using Time-Frequency Approaches: Coding, Classification, Fingerprinting, and Watermarking
Audio signals are information rich nonstationary signals that play an important role in our day-to-day communication, perception of environment, and entertainment. Due to its non-stationary nature, time- or frequency-only approaches are inadequate in analyzing these signals. A joint time-frequency (TF) approach would be a better choice to efficiently process these signals. In this digital era, compression, intelligent indexing for content-based retrieval, classification, and protection of digital audio content are few of the areas that encapsulate a majority of the audio signal processing applications. In this paper, we present a comprehensive array of TF methodologies that successfully address applications in all of the above mentioned areas. A TF-based audio coding scheme with novel psychoacoustics model, music classification, audio classification of environmental sounds, audio fingerprinting, and audio watermarking will be presented to demonstrate the advantages of using time-frequency approaches in analyzing and extracting information from audio signals.</p
Wavelet methods in speech recognition
In this thesis, novel wavelet techniques are developed to improve parametrization of
speech signals prior to classification. It is shown that non-linear operations carried out
in the wavelet domain improve the performance of a speech classifier and consistently
outperform classical Fourier methods. This is because of the localised nature of the
wavelet, which captures correspondingly well-localised time-frequency features
within the speech signal. Furthermore, by taking advantage of the approximation
ability of wavelets, efficient representation of the non-stationarity inherent in speech
can be achieved in a relatively small number of expansion coefficients. This is an
attractive option when faced with the so-called 'Curse of Dimensionality' problem of
multivariate classifiers such as Linear Discriminant Analysis (LDA) or Artificial
Neural Networks (ANNs). Conventional time-frequency analysis methods such as the
Discrete Fourier Transform either miss irregular signal structures and transients due to
spectral smearing or require a large number of coefficients to represent such
characteristics efficiently. Wavelet theory offers an alternative insight in the
representation of these types of signals.
As an extension to the standard wavelet transform, adaptive libraries of wavelet and
cosine packets are introduced which increase the flexibility of the transform. This
approach is observed to be yet more suitable for the highly variable nature of speech
signals in that it results in a time-frequency sampled grid that is well adapted to
irregularities and transients. They result in a corresponding reduction in the
misclassification rate of the recognition system. However, this is necessarily at the
expense of added computing time.
Finally, a framework based on adaptive time-frequency libraries is developed which
invokes the final classifier to choose the nature of the resolution for a given
classification problem. The classifier then performs dimensionaIity reduction on the
transformed signal by choosing the top few features based on their discriminant power. This approach is compared and contrasted to an existing discriminant wavelet
feature extractor.
The overall conclusions of the thesis are that wavelets and their relatives are capable
of extracting useful features for speech classification problems. The use of adaptive
wavelet transforms provides the flexibility within which powerful feature extractors
can be designed for these types of application
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