Brent Brown wrote:
> <snipped>
> Back to the subject, to digitally amplify a .wav file you would need
> to do something like this for each 8 bit sample:
> - Remove the offset by subtracting 7Fh for data above 7Fh or
> subtracting from 7Fh for data below 7Fh
> - Multiply by your amplification factor
> - Put the offset back in by adding 7Fh or subtracting from 7Fh as
> necessary
> - Store it or play it
>
> You are limited to what you can do in an 8 bit file before clipping.
> Using 16 bit intermediate maths helps by letting you do stuff like
> this: for 10% increase in volume, multiply by 11 then divide by 10.
>
> Brent Brown

Sound isn't linear, as perceived by the human ear, though;  We hear sort
of Logarithmically.  And you need to consider, what happens if you hit
overflow?, and handle that <G>

Really good books on Audio Accoustics are out there;  The average human
ear perceives a doubling in volume when the power has been increased by
10db, i.e. twice the actual power - so basically for a 10% increase in
*perceived* volume, you want to increase a signal's voltage by 10
TIMES.  So for a 10% increase in volume, I'd guess you actually want to
increase voltage by about 3.548 times?  (It's been a while since I did
audio engineering, and all my books are in storage due to the move.  I
like Digital lots more <G>)  So I'm possibly wrong here.

  Mark