On a sunny day (Mon, 10 May 2010 12:32:38 -0700) it happened Joerg wrote in :
VCO,
Well, if that takes so much time, say more then 2 hours, I think you need more time to go shopping or order the soundcard thing. Plus put it in a PC, plus write the soft, plus it is not real time adjustable with a pot, plus it is not GREEN, it sucks power, plus well it is all so obvious. Who knows, you may not reach your pension... may as well build it now!
Beats buying a 12-bit, 4 GHz dac! [1]
A 4:1 ratio, half of Nyquist, doesn't require heroic filtering. Above there the filter cost increases without limit as you approach 2:1.
John
[1] which actually exists
Some manufacturers make 4-6 channel codecs suitable for sound cards and home theater systems. These are typically specifies for frequencies above 3 Hz.
How low does a typical COTS stereo amplifier go ? While the amplitude response may be bad, why would the phase shift characteristics suffer from this ?
The cap tolerances would indeed cause problems.
For low production run devices, it would be perfectly feasible to use production measurements and software to compensate for the capacitor variations.
In discussing, which is the "best" way of implementing something, you always must consider the expected size of the production run. It is quite different to design a one off product or design a product for
100, 10,000 or 1,000,000 unit production / year, each requiring a different approach.
VCO,
obvious.
Hint: One can load revolving wave tables into a sound card :-)
-- Regards, Joerg http://www.analogconsultants.com/ "gmail" domain blocked because of excessive spam. Use another domain or send PM.
The one I use for testing small power supplies for international power standards or 400Hz aircraft supplies goes to under 10Hz. My sound card on the lab bench goes to 3Hz where it's 6dB down.
Yup, that's what messes up the phase down there.
It's temperature and often drive voltage dependent, for example if they used a Z5U cheapie in there. You can't tell even when looking at it.
Sure, that's why cost thinking has to go all the way, not just parts and labor.
-- Regards, Joerg http://www.analogconsultants.com/ "gmail" domain blocked because of excessive spam. Use another domain or send PM.
What about a single card with 4.1 or 5.1 outputs?
Either way, how would I generate the three 120 degree offset channels in software?
Thank you for your reply.
Bill Murphy
as VCO,
st.
If you are going to a digital approach, programming a cordic would be my advice. In the pre-internet days, the cordic was a great secret bit of code stolen from the early scientific calculators, and who knows where they got it. Nowadays it is all over the internet. The cordic was the heart of many a modem. We did one where the cordic was used in DTMF dialer. Needless to say it set a new standard in DTMF due to the quality of generating twin tones via DSP. Of course, it was a quality level that wasn't needed, but once you have a cordic programmed, you might as well use it. We even used the cordic in the FSK fallback, both in generating the FSK and in demod. For demod, the incoming signal was sampled and phrase unwrapped via the cordic, then the ramp (unwrapped phase versus time) was fitted via least mean squares to get the slope, which in turn yielded the incoming frequency.
Sitting over in Terman is a really great Phd dissertation on the cordic, better than any book I every read regarding the algorithm. My recollection is the author's name is Ahmed. Searching Stanford's Socrates doesn't seem to dig it up though.
What are the two other channels making?
If you have two signals at 90 degrees, you can get any other phase and the same amplitude.
There is a circuit that will do it, but it isn't electronic. It's a motor driving two AC generators (permanent magnet stepping motors are suitable) with angle adjustment for the stators to get to exactly the 120 degree shift you want.
If you want the audio range, an AIFF file with sine on left channel and cosine on the right channel is possible (but that doesn't have a continuous frequency adjust knob). The
120 degree shift is a linear combination of sine and cosine waves (look at the angle-sum formula). Your iPod or a sound card can reproduce the (audio-range) waveforms from the file, of course.The so-called all-pass Hilbert filter is not generally a practical project to cobble together in an afternoon. What one CAN do, is to make a square wave master clock, use flip-flops to generate slave clocks locked to the master, and then phase-lock sinewave generators to the slave clocks.
Another approach is to heterodyne. Mix up to some lowish IF, filter, mix back down. Apply phase shift at the IF to make the overall phase delay approximately 90 degrees. This works pretty well if the input frequency range (in octaves) isn't too wide, but it takes some care.
Cheers
Phil Hobbs
-- Dr Philip C D Hobbs Principal ElectroOptical Innovations 55 Orchard Rd Briarcliff Manor NY 10510 845-480-2058 hobbs at electrooptical dot net http://electrooptical.net
Perfectly OK, if the channels are truly discrete (no psychoacoustic matrix surround generation) and transparent (no band limiting in the .1 channel etc.).
You can precalculate a few (thousand) future samples into memory buffers (queue) and then let the sound card output the buffer at the speed specified by the sample clock. The buffers need to be updated, before the sound card has consumed all previous samples.
The actual sample generation is done in the same way as in DDS with a numerically controlled oscillator (NCO).
You need a 32 bit integer variable "phase accumulator" which is updated with a specific value at each iteration of a program loop, which defines the frequency. Take the high bits from the phase accumulator and use it to index a sine table. The value from the sine table is inserted into the queue going into the sound card (or written e.g. to a .WAV file).
To generate signals with a fixed phase relative to the master signal, take the current phase accumulator value, add a constant (the phase shift) and using the upper bits, access sine look up table and insert result into the queue for a different audio channel.
If the sample values are written into a .WAV file, the data can be replayed using any audio player.
The sine instruction is surprisingly fast on some x86 processors, so it could replace the sine look-up table. However, the phase accumulator must be an integer register, which overflows in a predictable way. A floating point register can not be used as a phase accumulator, since after long time, the least significant bits are lost and the sine function returns a constant value.
Same signal, all spearated by 120 degrees. Done in CoolEdit.
Can you give me a hint how, or where to find out please?
Bill Murphy
"Joerg"
** The use of 0.5% tolerance caps implies a phase ripple or error of better than 1 degree max.Using 1% tolerance or 1% values selected from 5% stock, the max phase error is not more than 2 degrees.
There is no problem designing a Hilbert phase shift network that covers from
22 Hz to 20 kHz using only 10nF and 1nF polystyrene caps of nominal 1% tolerance, 1% MF resistors and a few FET op-amps..... Phil
"Bill Murphy"
** Just by summing them in various ratios !!Call one " 0 " and the other " 90" - then summing equally gives 45 degrees relative to " 0" .
Invert the " 0 " signal - then summing in various ratios with " 90 " gives all the angles between 90 and 180 degrees.
Draw a phasor diagram and you will easily see it.
.... Phil
You should have said you wanted a tunable 3-phase supply (exciter).
0.1% resistors are cheap these days. Where can you buy PS caps of any tolerance ? NPO ceramic parts are available, but $$$$.
Best regards, Spehro Pefhany
-- "it's the network..." "The Journey is the reward" speff@interlog.com Info for manufacturers: http://www.trexon.com Embedded software/hardware/analog Info for designers: http://www.speff.com
"Spehro Pefhany" "Phil Allison"
** Still about 25 times the cost of 1% MF and of no benefit when the caps used are only 1% tolerance. ** LCR Components make polystyrene caps of 1% tolerance & Farnell / Newark sell them - among others.$1.54 each in *one off* cheap enough for you - eh ???
** Really ???1% tolerance * silvered mica * caps are available in values like 10nF for big bucks.
BTW:
What the HELL is the point of your TEDIOUS BLOODY nit picking ????
Got nothing better to do with your life ??
.... Phil
Should improve things by about 30%.
Cheap enough. I've heard a lot of complaints about poor availability of PS caps and PP caps.
Yeah, 18125A103FAT2A AVX 10nF 1% NPO in 1812 form factor, but about five clams a pop in tens.
Apparently, not at the moment, Phil. How about yourself?
Best regards, Spehro Pefhany
-- "it's the network..." "The Journey is the reward" speff@interlog.com Info for manufacturers: http://www.trexon.com Embedded software/hardware/analog Info for designers: http://www.speff.com
"Spehro Pefhany = PITA Wog Wanker "
** More nit picking hypotheses, not facts.Yawnnnnnnnnnnnnnn..............
Especially your own putrid variety.
** You could always do the planet a real favour a top your stupid self.Maybe just choke to death on your own bullshit.
Plenty of that on hand.
.... Phil
On a sunny day (Mon, 10 May 2010 13:57:33 -0700) it happened Joerg wrote in :
You can also load Mozart's into it :-) Or 'All you need is love' from the Beatles :-)
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