Modeling nuisance variabilities with factor analysis for GMM-based audio pattern classification

  • Authors:
  • Driss Matrouf;Florian Verdet;Mickaël Rouvier;Jean-François Bonastre;Georges Linarès

  • Affiliations:
  • University of Avignon, Laboratoire Informatique d'Avignon, 84911 Avignon, France;University of Avignon, Laboratoire Informatique d'Avignon, 84911 Avignon, France and University of Fribourg, Department of Informatics, 1700 Fribourg, Switzerland;University of Avignon, Laboratoire Informatique d'Avignon, 84911 Avignon, France;University of Avignon, Laboratoire Informatique d'Avignon, 84911 Avignon, France and Institut Universitaire de France, France;University of Avignon, Laboratoire Informatique d'Avignon, 84911 Avignon, France

  • Venue:
  • Computer Speech and Language
  • Year:
  • 2011

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Abstract

Abstract: Audio pattern classification represents a particular statistical classification task and includes, for example, speaker recognition, language recognition, emotion recognition, speech recognition and, recently, video genre classification. The feature being used in all these tasks is generally based on a short-term cepstral representation. The cepstral vectors contain at the same time useful information and nuisance variability, which are difficult to separate in this domain. Recently, in the context of GMM-based recognizers, a novel approach using a Factor Analysis (FA) paradigm has been proposed for decomposing the target model into a useful information component and a session variability component. This approach is called Joint Factor Analysis (JFA), since it models jointly the nuisance variability and the useful information, using the FA statistical method. The JFA approach has even been combined with Support Vector Machines, known for their discriminative power. In this article, we successfully apply this paradigm to three automatic audio processing applications: speaker verification, language recognition and video genre classification. This is done by applying the same process and using the same free software toolkit. We will show that this approach allows for a relative error reduction of over 50% in all the aforementioned audio processing tasks.