18 Feb
2011
18 Feb
'11
8:02 p.m.
On Fri, Feb 18, 2011 at 07:53:46PM +0200, Alfs Kurmis wrote:
Because the theory says so. You can either
1. believe me,
2. study the theory yourself,
3. try it out yourself.
2 and 3 would be the best thing to do. See also the example
at the end of this post.
It  means block schema of Automatick
Gain/Volume Control is
(can be)
Please try to avoid those   characters in your mails....
Filter -- RMS_Calculator -- Volume_control 
In exact such order ?
Yes. The filter is not essential, you can get good results
without it. Most compresssors or AGCs don't have a filter.
(- For example in FFT are used Rectangular(no window)
, Hann,
Hamming, Barlow ... windows. -)
FFT windows have nothing to do with this.
What are the most common filters for ACG and
"Loudness" control ?
Just a simple first order lowpass acting on the squared samples.
Nothing
special is needed for high frequencies.
Why not ? So far i unterstand the best way for RMS calc would
be SQRT of
integral of power2 of sound signal function.
Real signal is not sequence of sampled rectangles, but smooth
function.
Can not happen so what that rectangle inaccuracy of each sample by
freq > 10KHZ ,
in end effect will accumulate big inaccuracy ?
The analog signal is *not* the samples converted to rectangles
and smoothed a bit. It looks like that for low frequencies, but
what is really happening in DA conversion is something completely
different.
U mean that normally full amplitude sine wave is
defined as 0dB RMS
signal ?
In most cases that is the definition of '0 dB'.
Can U plz gimme some examples ?
Take a sine wave with peak amplitude +/- 1. The samples are:
sample [i] = sin (w * i)
i = sample number.
w = 2 * pi * frequency / sample_frequency.
Now the square of sin(x) is 0.5 + 0.5 * cos(2 * x)
The average value of cos(2 * x) is zero, so the average
value of the square of sin(x) is 0.5, and the RMS value
is sqrt(0.5) =~ 0.7071.
It doesn't matter where the samples are: if there are enough
of them then the average of cos(2 * x) will be zero, and the
result of the RMS calculation will be sqrt(0.5). ** Also for
high frequencies. **
A square wave of amplitude +/- 1 has RMS value 1. So if you
use the sine wave above as the reference (0 dB) then the
square wave is +3 dB.
But note that if you sample an analog square wave, the samples
will in most case *not* be +/- some single value. And if your
samples are +/- some value, then the analog waveform will in
most cases *not* be a lowpassed square wave (in both cases
it will be close at low frequencies). The relation between
samples and the analog waveform is not as simple as that.
Ciao,
--
FA