I read in sci.electronics.design that Ken Smith wrote (in ) about '"all pass" thought about (analogue) compression', on Fri, 7 Jan
2005:
I don't immediately see how that would work for a broadband input signal. Splitting the signal into octave bands and processing as you propose would indeed work, because the third and higher harmonics are out-of-band, if the all-pass maintains its 180 degree phase-shift, relative to that at f to 2f, from 3f to 6f, where f is the lower band- edge frequency of an octave-band filter.
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Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
Imagine that we have broken the clipping operation into two steps. Further imagine that we have made these steps such that if the first step adds NmV of 5th harmonic to the signal, the second does also. This means that the second step is a little harder than the first.
For purposes of thinking about it assume, we first pass the signal through just the clippers with no phase shifter between them and record the spectrum of the result. Then we do this:
An all pass filter with a modest Q can shift, lets say, the 1KHz to 5Hz band. The phase curve suddenly starts adding delay at about the
1KHz point.
Any harmonic, made from a signal well below 1KHz, that is above the 1KHz point will be shifted in phase relative to its fundamental.
If this signal is again clipped, new harmonic components will be created in the clipping process. These new components will be at some phase angle to the shifted ones that have passed through the all pass filter.
The sum of two vectors is at its maximum when the vectors are aligned. Any phase difference between the new harmonics and the ones from the all pass means that the amplitude of the sum will be less than if there was no phase shift.
Over some band of frequencies, the phase shift will be between 120 and 240 degrees and the harmonics will tend to cancel.
Since none of the harmonics can be greater than the case where there was no shifter but some are smaller, the THD is less for the circuit with the phase shifter.
That sounded clear to me, but I already know what I was thinking.
I read in sci.electronics.design that Lasse Langwadt Christensen wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Fri, 7 Jan
2005:
Loudspeakers are voltage-operated, in the sense that they give the designed frequency response with constant-voltage input, not constant power input. In any case, a series element can't divert any current; it can only reduce current as a consequence of developing a voltage across itself.
Yes. A very long, thin filament in a gas-filled lamp might have a thermal time-constant for fairly small temperature rise low enough not to matter, but that's very low; less than 10 ms.
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Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
I have a pair of speakers that use a lamp it to protect the tweeters, but I'd say its more of a power limiter than a voltage limiter and wouldn't it violate your requirement of not having a time constant to worry about ?
I read in sci.electronics.design that Ken Smith wrote (in ) about '"all pass" thought about (analogue) compression', on Fri, 7 Jan
2005:
It sounds clear to me, as well, and I expect it would work, at least to some extent.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
Yes, the quietist to loudest range is likely to be about 60dB or so, if that is sense in which you mean dynamic range. But at the low end (when someone wispers), we still need a few bits in the ADC. 16 bits would most likely work if we oversampled and could count on noise to smear out the artifacts at low amplitudes. It ends up being a trade off between bits and speed.
I read in sci.electronics.design that John Larkin wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Fri, 7 Jan
2005:
Yes, this sort of thing is kept proprietary and heavily protected by patents.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
I read in sci.electronics.design that Fred Bartoli wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Mon, 10 Jan 2005:
I've got something similar to that at present, although it has *bass- cut* pre-processing rather than treble-boost. Jim Thompson's tanh limiter does seem to have advantages, but I'll try your pre-processing first, because that involves fewer changes to my breadboard.
I planned to do some work on it today, but of course, those damned 'clients' have intervened. (:-(.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
That's a general problem where I can't see how you've provided a method. How do you propose a system resolve two tones and raise power individually. If this were possible, we really could have amplifiers with *no* IM products. It has never to my knowledge been accomplished and really seems a contradiction of terms: non-linearity that has no non-linearity products. ???
The input to the system is random. "Tone frequencies" are unknown beforehand. More specifically, the inputs are not even tones, and no reasonable filtering method could have the needed resolution. The actual implementation of a (·)^(m/n) *frequency domain* "clipper" needs a bit more discussion too.
Okay, lets define a standard two-tone equal level signal (let's use analytic signals for ease):
x(t) = (e^(j·w1·t) + e^(j·w2·t))/(2*pi)
These tones can be arbitrarily "close" or "far" apart.
FT'ing this:
F{x(t)} = X(jw) = dirDel(w-w1) + dirDel(w-w2)
where
dirDel(·) := the dirac delta function
Because applying the rational power function to each tone individually seems to have no obvious general solution, we apply it to the input generally:
Y(jw) = (dirDel(w-w1) + dirDel(w-w2))^(m/n)
where m/n is some rational fraction; 5/7 if you like.
I don't care if you evaluate in the frequency or time domain. It remains to be shown how this will not produce distortion products, either IM or harmonic (harmonic is simply a subset of IM anyway.) I would actually like to see the transform of the integral anyway.
"The intermodulation distortion will not be made zero by this method. If the input has more than one frequency component, the distortion will be much higher."
So unless I misunderstand you, there is a contradiction.
You run the signal into a DSP that does what I suggested. This forces you to use a DFT but for material of limited length, the DFT is good enough. The system has a very large input to output delay but it does not have to produce IM distortion. This is not something that can be part of a realtime system since the input to output delay is greater than the material length. It is, however, something that could be done to the material on a CD and a new CD written to be played at a later time.
[...]
That sentence is from a different context about a completely different subject.
Yes, you've mix up two ideas that are not related.
Wierd!. Never had chance to play with a compressor, so ran a 2 second 8kb .WAV through the above (an explosion) and listened to the result. What surprised me was the output waveform had been compressed, yet every nuance of the original waveform followed through to the output. Ran the .WAV through 2 of the above circuits in series (X2 gain in between). Even more compression even more of a 'levelled' explosion. Think I'll stick 8 in series and listen to what happens. Maybe salt in some bass cut varieties!. regards john
I've only a few *.WAVs on the PC, so I'll see if I can pull something from the net. I'll post the (single unit) before and after WAVs to ABSE. (Input WAV is 8bit at 8k bits per second. Output WAV saved as 8 bits at
11kbits per second. Single channel only. (Ps, Done via an LTSpice sim, so the diodes suffer perfectly symmetry) regards john
"john jardine" a écrit dans le message de news:crusfo$pt4$ snipped-for-privacy@news7.svr.pol.co.uk...
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I didn't tried this but the sims let me expect something like this. Can you run some other program through it, like speech and music ?
Maybe you have the opportunity to make a wav and post it back ? I'd be curious.
For the simulation, run a 500Hz + 4kHz sin waves at various levels, and observe the output, vs a simple R+diode clipper. This one is amazing and can handle really **huge** overload levels. (I didn't tried more than 2 sinus)
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