🚀 go-pugleaf

I recently tackled the sample-rate conversion problem in some of my
own work.  The paper that got me on the right track is:

"Interpolation and Decimation of Digital Signals - A Tutorial Review"
by Ronald E. Crochiere and Lawrence R. Rabiner.
This paper is in Proceedings of the IEEE, Vol. 69, No. 3, 1981, pp.
300-331.

In short, efficient implementation of Sample Rate Conversion involves
filtering short windows (of length Q) of the input data using a
polyphase filter (FIR with Q taps) operating at the output sample
rate.  The polyphase filter has periodically time-varying
coefficients.

The sample rate converter should be quite do-able on a DSP Chip.  Use
an up-conversion factor of 160 and a down-conversion factor of 147.
(44.1 * 160/147 = 48)  The polyphase filter cycles through 160 states,
then repeats.  The filter coefficients would be stored in 160*Q words.
For each output sample, besides managing windowing of the input data,
just Q multiply and accumulates need to be performed.  Typical values
of Q range from 5 to 20.

As far as determining the filter coefficients goes, read the paper!

I have implemented a sample rate converter along these lines in
Matlab.  It is good enough for my work, but is too slow for signals
greater than a few thousand samples in length.

				Brent Weeks
				bweeks@fermat.uvic.ca