On Fri, Feb 18, 2011 at 07:53:46PM +0200, Alfs Kurmis wrote: > 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 ? 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. > 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 _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
