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SOURCE-SIGNAL ANALYSIS 573<br />

and some, such as the violin, can have significant antiresonant valleys as<br />

well.<br />

In technical terms, the resonant peaks are called poles <strong>of</strong> the system<br />

function while the notches, if present, are termed zeroes. In speech and<br />

music work, the zeroes are frequently ignored and the poles are called formants.<br />

The goal <strong>of</strong> formant estimation is determination <strong>of</strong> the number <strong>of</strong><br />

formants, their resonant frequencies, and possibly their bandwidth or Q<br />

factors. This information represents the characteristics <strong>of</strong> the filter, which is<br />

usually a majority portion <strong>of</strong> the analyzed sound's timbre. Musical applications<br />

<strong>of</strong> formant analysis seek the design <strong>of</strong> a filter with the same formant<br />

structure as the analyzed sound. Different synthesized excitation functions<br />

are then modified by the filter. The resulting sound has the most prominent<br />

characteristics <strong>of</strong> both. For example, if vocal vowel sounds are used for the<br />

formant analysis and the resultant filter is driven with a square-wave excitation<br />

function, the vowel quality would be retained but with the characteristically<br />

hollow timbre <strong>of</strong> square waves!<br />

Perhaps the most straightforward method <strong>of</strong> formant analysis is to scan<br />

the amplitude spectrum and find the center points <strong>of</strong> high-amplitude clusters<br />

<strong>of</strong> harmonics. The center points are the formant frequencies. The bandwidths<br />

may be estimated by noting the 3-dB points on either side <strong>of</strong>each peak. This<br />

method works well only if the harmonics are dense compared to the<br />

bandwidths <strong>of</strong> the formants, which means a low excitation frequency. This is<br />

because the excitation function harmonics effectively sample the filter response<br />

curve and insufficiently dense sampling leads to aliasing and incorrect<br />

conclusions regarding the response curve. Even if the sampling is theoretically<br />

dense enough, interpolation may have to be used to increase it further so<br />

that the center point and 3-dB points <strong>of</strong> the peaks can be accurately determined.<br />

Note that the original spectral analysis need not resolve the harmonics<br />

themselves to be useful for this method <strong>of</strong> formant analysis.<br />

Linear Prediction<br />

A more refined method <strong>of</strong> formant analysis is called linear prediction<br />

because, by specifying the filter involved in the original sound generation,<br />

one is able to predict what future time samples <strong>of</strong> the sound are likely to be.<br />

Although originally developed for the efficient transmission <strong>of</strong> speech signals<br />

over limited bandwidth channels, linear prediction is useful in music synthesis<br />

because it results in an actual design <strong>of</strong> the filter represented by the system<br />

function. Unforrunately, the calculations are too involved to be covered here<br />

in simple terms but the general characteristics <strong>of</strong> linear prediction can be<br />

discussed. Additional reference material is listed in the bibliography.<br />

In linear prediction, the actual filter used to generate the sound being<br />

analyzed is approximated by a filter having a specified number <strong>of</strong> poles and<br />

zeroes. Often, because <strong>of</strong> computational difficulties encountered in handling

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