On Wed, Jul 21, 2010 at 11:52 AM, JohnLM <johnlm@email-addr-hidden> wrote:
> Hey, fellow devs!
> I'm a newcomer to this mailing list and I'm sure this has been discussed
> here at some point, but I'll ask anyway.
>
> I'm a bit better on programming side than on audio-tech side so...
> - Does all floating point formats mandate valid values in [-1.0, 1.0]
> range?
most systems that use floating point to represent linear PCM values
normalize to this range, yes. that doesn't mean that values outside
the range are invalid, but when converted back to integer values they
will clip.
> - How does floating point (and perhaps fixed point/integer) values relate
> to dB notation?
floating point -1.0 == minimum value of an integer sample == -inf dBFS
floating point 1.0 == maximum value of an integer sample == 0dBFS
relationships to dBU, dBV etc are not fixed.
> - If I muliply float by 2.0, do I get twice the sound pressure or twice the
> sound intensity, or something else?
you get samples that are twice as loud. this doesn't have a linear
relationship with sound pressure or intensity or perceived volume
other than "its louder"
> Well as for internal processing. If I mostly use integer formats for inputs,
> it doesn't appear I gain anything from "normalizing" them (scale to [-1.0,
> 1.0]) during internal processing. I just output integer part directly to
> integer formats (adjusted for different bit resolutions when necessary), and
> normalize only when saving to float format. What do you think?
the reason you gain from using floats is that you will not clip when
adding or scaling values. this is a huge win.
> Proprietary formats such as AC3 and DTS market themselves as high dynamic
> range. Do they use values over 0dBFS or do they just have higher bit
> precision than "standard" formats?
neither. they are compression schemes, nothing else.
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Received on Thu Jul 22 00:15:01 2010
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