Re: audio sampling rate question
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?
-- glen
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