On Tue, 06 May 2008 20:51:45 -0800, glen herrmannsfeldt
<[email protected]> wrote:

>Ben Bradley wrote:
>(snip, I wrote)
>
>>>I don't know about the OP, I was recently using a Roland R1
>>>recorder. It has the choice of 16 or 24 bit WAV, and some
>>>choices of MP3. With 24 bits, I then wrote a C program to
>>>find the minimum and maximum sample values, along with a some
>>>other statistical values. I then did the conversion to 16 bits
>>>by rounding and shifting the appropriate number of bits.
>
>> This doesn't sound right. If you want the best quality possible in
>> the final 16-bit signal, look into dithering and noise-shaping rather
>> than doing rounding, and it's best to do all your processing in
>> floating point (that way you can change the level to any arbitrary
>> value, rather than only in the 3dB steps that bit shifting gives).
>> OTOH, the difference can be subtle, and many people can't tell even
>> under the best listening conditions.
>
>I thought about it, but so far I haven't tried. By finding the
>peak and appropriate shifting, the quietest parts aren't so far down.
>
>It is a recording with a live audience, and the background isn't all
>that quiet, anyway. It would be nice, though. Do others do it?

Yes, I'm pretty sure most all audio editing software (everything
from Audacity to Pro Tools) have been doing it as I described for many
years now.
It might be easier to write some script file for an audio editing
program to do what you want. I don't know what programs have what
features nowadays, but the old Cool Edit 2000 has/had scripting so you
could do several automated things with audio files.

>
>-- glen