🚀 go-pugleaf

In the digital audio world, a lot of sound is recorded at a 48k sample
rate. A CD, however, is at 44.1k. As we all know, the classic
technique of sample rate conversion would have us upsample to the
lowest common multiple, low-pass filter, and downsample to the desired
rate. The only problem is that the LCM of 44.1k and 48k is 21.168 Meg!
A 4 minute stereo soundfile at 48k takes up about 50 megabytes, which
means that it's 21.168M upsampled version would be over 22 gigabytes!

Obviously, sample rate converters don't store the entire intermediate,
upsampled version, but I was wondering if anyone has a good idea of 
what sorts of schemes they do use. Do they do some sort of overlap-
save method akin to block convolution? Does anyone have any code
to do this?

- Joe Chung