I remember this table now... Forgot where it was, to be honest. I'll Link the alt.binaries.sounds.mods FAQ v2.6 directly:
Part 1 /
Part 2Ypu've likely figured it all out, but I'll re-post and explain what is going on. The following is a list of standard notes in a scale, and their frequencies in Hz
(supplied by Chris Craig, author of GoldWave)
C - 261.7 E - 329.6 G# - 415.3
C# - 277.2 F - 349.2 A - 440.0
D - 293.7 F# - 370.0 A# - 466.2
D# - 311.1 G - 392.0 B - 493.9
Bear in mind that doubling the sampling rate raises a sound by one octave,
halving the rate lowers the note by one octave.
Here's the math, put in simple terms that a person can understand:
Old Sample rate / Old note Frequency * New Note Frequency = New Sample Rate
This makes perfect sense to most people. Take a sample, split it into it's smallest part (1 Hz) then multiply it by the # of Hz you want it to be....
To a computer that is doing integer math, doing the division first could cause your program/calculation to lose accuracy... As these are all multiplication/division, you can perform the operations in any order. Thus rewriting it as:
Old Sample Rate * New Note Frequency / Old Note frequency = New Sample Rate
works better and causes less data loss through truncation (not that it'll make a huge difference with larger sample rates)
Most Samplers these days have transposing effects that do the math for you, and resample it to whatever the sample rate is set at, without having to set the base frequency inside the tracker (After Windows 98 got installed on my PC, I used Goldwave to convert the sample altogether right up until Audacity came out). You're trying to do it the old-school efficient way though. Which is equally effective, but requires you to get mathematical. Most of us just got lazy.... or decided to sample Middle C instead
