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Abstract

Audio sample rates are converted by an arbitrary ratio of Q/P using a two-stage sample-rate converter. One stage is an L-tap low-pass finite-impulse-response (FIR) filter, while the other stage is a linear interpolator. Coefficient storage for the L-tap low-pass FIR filter is dramatically reduced by reducing the effective P factor. The effective P factor is reduced by using two stages, with each stage adjusting the sampling rate by a different ratio. A first stage adjusts the sampling rate by Q0/P0, while a second stage further adjusts the sampling rate by Q1/P1. Q0 and P0 are large integers of about 400 to 700 that differ by one or three; thus the ratio Q0/P0 is very close to one. The linear interpolator stage eliminates or adds one or three samples and smoothes the samples by linear interpolation over the 400 to 700 remaining samples. The FIR filter stage adjusts the sample rate by a ratio of Q1/P1, which is approximately but not exactly Q/P. The FIR filter smoothes the expanded or reduced samples using weighting coefficients. Only P1 sets of coefficients are stored, rather than P sets. Since P1 is only about 10, while P is 400 to 700, coefficient storage is dramatically reduced.

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