I read in sci.electronics.design that John Larkin wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Wed, 5 Jan
2005:True, this technique is well-known, but it's costly. I'm looking for an ingenious low-cost solution.
I can do that with a -2 ohm resistor, which has a conductance of -half a mho.
-- 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
Come on now John, are you saying you know what you want more than we do ?:-)
...Jim Thompson
--
| James E.Thompson, P.E. | mens |
| Analog Innovations, Inc. | et |
| Analog/Mixed-Signal ASIC\'s and Discrete Systems | manus |
| Phoenix, Arizona Voice:(480)460-2350 | |
| E-mail Address at Website Fax:(480)460-2142 | Brass Rat |
| http://www.analog-innovations.com | 1962 |
I love to cook with wine. Sometimes I even put it in the food.
I'll come back to the file at this link...
Any VCA-based (or "opto" [LED/Light Bulb and light-dependent resistor] or vacuum-tube based "Vari-MU") compressor or limiter will "involve at least one time constant" but the definition of a "limiter" is (in addition to a near-infinite compression ratio) that the attack time is short enough to be negligible and no peak will come through.
You didn't object to the circuitry in the first link (AN176.pdf) not being instantaneous, and for the circuits described, it's clearly not. Quoting from page 10-6:
"CRECT acts as the rectifier?s filter cap and directly affects the response time of the circuit. There is a trade-off, though, between fast attack and decay times and distortion."
It doesn't take too much circuitry to make the attack and decay times independent (little more than an op-amp as voltage follower) and have the attack time arbitrarily short, though of course a fast attack will always distort the first wave at the onset of a louder signal, as the gain is reduced as the instantaneous input signal goes above the threshold. This might make a 'click' as the first quarter-cycle or so is flattened, but I don't think it should be too audible or objectionable.*
You might ask this on rec.audio.pro where they use this sort of stuff every day.
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I read in sci.electronics.design that Ben Bradley wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Thu,
6 Jan 2005:I'm not interested in an argument. I may not have commented on that point, but it's an overall requirement, as shown by the word 'instantaneous' in the Subject line.
-- 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
National Semiconductor has a good audio agc circuit for their dual transconductance amp (LM13700). How cheap is cheap for you? The next cheapest thing I could think of would be to use a jfet as a voltage controlled resistor, however for that to work without an amplifier you would need a fairly large amplitude audio signal to begin with as the peaks would determine the amount of resistance in the jfet.
I read in sci.electronics.design that Jim Thompson wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Thu, 6 Jan 2005:
How could I possibly be so presumptuous?
-- 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
Can you stand a little delay in the output signal? I'm thinking in terms of a mS or so. Earlier, I suggested something like a slew rate limit to supress the high frequency components of the result. I have a 1/2 formed idea that may do a bit better based on using a couple of "all pass filters".
I'm at work waiting for a sim. to finish so thinking it through will have to wait for tonight.
-- -- kensmith@rahul.net forging knowledge
I read in sci.electronics.design that Ken Smith wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Thu,
6 Jan 2005:Yes. In fact, if you could make two channels, one having 10 ms more delay than the other, I could use that for another useful purpose.
-- 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
If you take a sine wave and run it through a circuit that does:
Y = X ^(17/19)
the sine wave's RMS amplitude will be compressed towards about 0.98V RMS and there will be some distortion. The 3rd harmonic will be about 2.7%.
Assume that the sine wave we start with is 300Hz.
A phase shifter (all pass filter) can be made with a Q such that the
900Hz, 3rd harmonic is shifted by 180 degree relative to the 300Hz sinewave.If we take this shifted signal and do another X^(17/19) operation on it, the 3rd harmonic will only be about 0.2%
You don't need the phase shift to be exactly 180 degrees. Any non-zero phase shift and two steps of (17/19) soft clipping will result in less harmonic content than one step of (17/19)^2 clipping would produce.
If more distortion can be lived with, a lower power such as (11/13) could be used.
Since the band of interest is 300Hz to 3KHz, we don't have to worry about the harmonics of the frequencies above 1KHz. Those can be removed with a simple low pass filter. I haven't verified it yet but it seems to me that
3 stages of phase shifter and 4 clippers should be able to make a significant compression of amplitude but make less that 5% distortion on a sine wave.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.
-- -- kensmith@rahul.net forging knowledge
Absurd, I think you should look how a sine wave changes its values, it gets zero, then negative, try it then. How do you want to compress, analog or digital? and how do you want to get the envelop signal. Which time constants?
Tell me which drugs are you using? Better use a u-law A/D or something analog like THAT4301(better than 0.1%).
-- ciao Ban Bordighera, Italy
Sounds ideal for a cheap DSP chip.
John
I read in sci.electronics.design that Ken Smith wrote (in ) about '"all pass" thought about (analogue) compression', on Fri, 7 Jan
2005:
It isn't: I've posted that it's 100 Hz to 5 kHz (at least).
Interesting, a bit more ambitious circuit-wise than I really expected, and how to realise the weird fractional power law?
The input is mainly speech, but there could be music as well. In any case, many frequencies.
-- 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
The first lab prototype that will be built next month, will go zonkers, thus producing that Richter 9 earthquake..
[snip]
Say you hard low-pass before 10KHz, then an all-pass at 10KHz will give 100us of delay.
I've used this scheme to process audio to eliminate "pops" from records.
But I'm unsure of any value for your compression needs.
...Jim Thompson
--
| James E.Thompson, P.E. | mens |
| Analog Innovations, Inc. | et |
| Analog/Mixed-Signal ASIC\'s and Discrete Systems | manus |
| Phoenix, Arizona Voice:(480)460-2350 | |
| E-mail Address at Website Fax:(480)460-2142 | Brass Rat |
| http://www.analog-innovations.com | 1962 |
I love to cook with wine. Sometimes I even put it in the food.
ISTR the Datong RF clipper from the 70's. that gave a 6dB improvement. I've done a quick google, but nobody seems to have the circuit
martin
Serious error. All shortcuts have disappeared. Screen. Mind. Both are blank.
I read in sci.electronics.design that martin griffith wrote (in ) about 'Instantaneous (analogue) compression of speech signals', on Fri, 7 Jan 2005:
It uses SSB clipping. It's far too complicated for what I want.
-- 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
In article , John Larkin wrote: [... DSP based AGC ...]
For one channel of voice grade signal, I'd bet a PIC or 8051 based circuit could do it. The tricky bit is the dynamic range of the ADC. It is easy to get 24bits worth of analog dynamic range and harder to get that in an ADC.
-- -- kensmith@rahul.net forging knowledge
This 'delay' thing is another issue entirely. You can get useful field patterns by deploying two or more loops carrying uncorrelated speech signals. One method of decorrelation is to use a wide-band 90 degree phase-shift and we've already discussed my AFILS phase-shifter here. Another method is to use a delay of around 10 ms. This isn't as nice as it might seem, because it gives a comb-filtered frequency response, and some people seem to be able to hear it. I couldn't, even before I went deaf.
You mean how much compression? Well, that depends on the subjective evaluation. Too much gives a very penetrating sound, that is quite unpleasant.
I wasn't able to do the subjective test with my colleagues today; we ran out of time discussing speech intelligibility measurements!
-- 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
You're not likely to see much more dynamic range than 60 or so dB for any real-world audio signal. So a 12-16 bit ADC should be good for most apps. A DSP, or even a decent uP, could delay the data stream, do an average or quasi-peak detection, envelope delay that some clever smooth way, and multiply the delayed samples to compress the dynamic range without bad artifacts. I'm sure it's being done already.
John
That would make the circuit need a few more sections.
I'd aproximate it with a it of curve fitting. Perhaps the sum of a few long tail pairs.
The "more ambititious" than expected problem could be, I think, the killer for this idea.
Perhaps there is still something in the idea worth considering. Whatever clipping curve you apply to the signal could be broken into 2 parts and a simple all pass filter used. The result should be no worse than the one stage of clipping and may in fact sound better. Instead of trying to zero the 3rd harmonic, higher harmonics could be targetted.
-- -- kensmith@rahul.net forging knowledge
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