Demonstrates how the sample-rate can be arbitrarily modified with downstream components automatically adjusting
themselves to the new sample rate.
In this example, three sample rates are present: the 44100 Hz base rate of the stereo audio file; the rate of 132300 Hz for
the processing of the left channel which as been up-sampled by a factor of 3; and the rate of 22050 Hz for the processing
of the right channel which has been down-sampled by a factor of 2. The 1-tap Delay blocks automatically configure
themselves such that the desired delays are achieved irrespective of the sample-rate.
Note that the re-sampling of the left branch is carried out using simple "sample and hold" methods, whereas the
re-sampling of the right branch is carried out using FIR half-band (Nyquist) polyphase filters. This is because, in this
example, the right channel initial down-sampling is more sensitive to aliasing/imaging than the left channel up-sampling
since the audio material contains mostly low frequencies in the audio range. Hence the right channel requires the use of
anti-aliasing / anti-imaging filters which are implicit in the Nyquist polyphase filters.
Try experimenting with different re-sampling blocks to appreciate the effects of aliasing and imaging.
Also note that both branches are ultimately brought back to the sample rate of 44100 Hz which is that supported by the
audio output device.
