5,655 research outputs found
A modulation property of time-frequency derivatives of filtered phase and its application to aperiodicity and fo estimation
We introduce a simple and linear SNR (strictly speaking, periodic to random
power ratio) estimator (0dB to 80dB without additional
calibration/linearization) for providing reliable descriptions of aperiodicity
in speech corpus. The main idea of this method is to estimate the background
random noise level without directly extracting the background noise. The
proposed method is applicable to a wide variety of time windowing functions
with very low sidelobe levels. The estimate combines the frequency derivative
and the time-frequency derivative of the mapping from filter center frequency
to the output instantaneous frequency. This procedure can replace the
periodicity detection and aperiodicity estimation subsystems of recently
introduced open source vocoder, YANG vocoder. Source code of MATLAB
implementation of this method will also be open sourced.Comment: 8 pages 9 figures, Submitted and accepted in Interspeech201
Probabilistic Modeling Paradigms for Audio Source Separation
This is the author's final version of the article, first published as E. Vincent, M. G. Jafari, S. A. Abdallah, M. D. Plumbley, M. E. Davies. Probabilistic Modeling Paradigms for Audio Source Separation. In W. Wang (Ed), Machine Audition: Principles, Algorithms and Systems. Chapter 7, pp. 162-185. IGI Global, 2011. ISBN 978-1-61520-919-4. DOI: 10.4018/978-1-61520-919-4.ch007file: VincentJafariAbdallahPD11-probabilistic.pdf:v\VincentJafariAbdallahPD11-probabilistic.pdf:PDF owner: markp timestamp: 2011.02.04file: VincentJafariAbdallahPD11-probabilistic.pdf:v\VincentJafariAbdallahPD11-probabilistic.pdf:PDF owner: markp timestamp: 2011.02.04Most sound scenes result from the superposition of several sources, which can be separately perceived and analyzed by human listeners. Source separation aims to provide machine listeners with similar skills by extracting the sounds of individual sources from a given scene. Existing separation systems operate either by emulating the human auditory system or by inferring the parameters of probabilistic sound models. In this chapter, the authors focus on the latter approach and provide a joint overview of established and recent models, including independent component analysis, local time-frequency models and spectral template-based models. They show that most models are instances of one of the following two general paradigms: linear modeling or variance modeling. They compare the merits of either paradigm and report objective performance figures. They also,conclude by discussing promising combinations of probabilistic priors and inference algorithms that could form the basis of future state-of-the-art systems
Spectral analysis for nonstationary audio
A new approach for the analysis of nonstationary signals is proposed, with a
focus on audio applications. Following earlier contributions, nonstationarity
is modeled via stationarity-breaking operators acting on Gaussian stationary
random signals. The focus is on time warping and amplitude modulation, and an
approximate maximum-likelihood approach based on suitable approximations in the
wavelet transform domain is developed. This paper provides theoretical analysis
of the approximations, and introduces JEFAS, a corresponding estimation
algorithm. The latter is tested and validated on synthetic as well as real
audio signal.Comment: IEEE/ACM Transactions on Audio, Speech and Language Processing,
Institute of Electrical and Electronics Engineers, In pres
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