  Introduction Several speech processing algorithms assume the signal is
stationary during short intervals (approximately 20 to 30 ms). This assumption
is valid for several applications, but it is too restrictive in some contexts.
This work investigates the application of adaptive signal processing to the
problem of estimating the formant frequencies of speech. Two algorithms were
implemented and tested. The first one is the conventional Least-Mean-Square
(LMS) algorithm, and the second is the conventional Recursive Least-Squares
(RLS) algorithm. The formant frequencies are the resonant frequencies of the
vocal tract. The speech is the result of the convolution between the excitation
and the vocal tract impulse response [Rabiner, 78], thus a kind of
"deconvolution" is required to recover the formants. This is not an easy
problem because one does not have the excitation signal available. There are
several algorithms for formant estimation [Rabiner, 78], [Snell, 93], [Laprie,
94
