I wanted to simulate what would happen if a recording was made and mixed
at 176.4kHz, down-sampled for a CDDA master using a steep linear phase
filter, and then upsampled in two different playback systems, one using
a steep linear phase filter and the other using a relaxed minimum phase
filter with a low pass corner at around 20kHz. All the filtering steps
were implemented using sox.
The 3 steps starting from an ideal impulse representing the original
recording were as follows:
Code:
--------------------
sox -D -V myimpulse24.wav -b 16 mydown.wav rate -vs 44100
sox -D -V mydown.wav -b 24 -t flac myupsteep.flac rate -vs 176400
sox -D -V mydown.wav -b 24 -t flac myupmp.flac rate -b 85 -Mam 176400
--------------------
The first step makes a 16-bit 44.1kHz version of the original
impulse/program material. The only way you could get to listen to this
version would be with a NOS DAC, but then you would have to tolerate the
resultant aliasing.
The second step does 4x upsampling using a steep linear phase filter.
Some CD players in production probably use similar filters.
The 3rd step postulates an alternative upsampling using a minimum phase
and rather relaxed filter. You can observe the elimination of the
pre-echo compared to the more conventional approach, when comparing the
respective cases in the figure below.
I think that the basic princple of the apodizing filter has been
demonstrated by these plots. Whether it sounds any good is another
question entirely, but one that some of our forum members might want to
answer for themeselves...
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--
dsdreamer
----------------------
"Dreamer, easy in the chair that really fits you..."
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