Re: a little oscillator theory

Which contain zero information, naturally. ;)

Cheers

Phil Hobbs (Now if _I_ had a hammer....)

Yes the AGC is very important for dual resonance oscillators, without it you tend to get oscillation at one or other of the two frequencies but not reliably both together.

In an audio intermodulation meter I designed, the two tone sources had to be as noise-free as possible. The problem was that any noise from the gain-determining element of the high-frequency oscillator intermodulated with the oscillator signal and appeared as if it were amplitude variations. That set a limit to the maximum sensitivity of the measurements.

After trying various configurations, I found that the quietest results came from a Wien Bridge type of circuit where the gain was accurately set to just over 3 and the op-amp was allowed to clip. The wavform and frequency stability weren't brilliant, but that didn't matter.

As a noise-free way of controlling the amplitude, I generated symmetrical low-noise power supply rails from op-amps driving emitter-follower buffers, cntrolled by the voltage from a 'set level' control on the front panel. Both oscillators were controlled by this method, so their amplitudes tracked together.

As I recall, that is exactly how it works. The drawing above came from the BSTJ article on the design, published in the 1950s.

It helped to rearrange the circuit diagram, simplifying the tone pad switching details.

I have the article somewhere, from when I was designing some oscillators in the 1970s, but have not seen it in decades.

Joe Gwinn

I'm thinking that my next oscillator will use an IC, specifically a BUF602, instead of a phemt as the gain element. It's a1 GHz unity-gain buffer that I assume TI has done right. It shouldn't oscillate at 6 GHz or whatever.

It still needs some controlled amplitude limiting, which I'll do with a diode. I sure don't want the BUF602 to rail to limit oscillation amplitude.

"Squegging" is an ill-defined term, Essentially it is a chaotic oscillation, which can repeat exactly but frequently doesn't.

Hard clipping always generates higher harmonics - mostly all of them up to a limit set the period in which each cycle is clipped. Those harmonics can excite other resonances.

I've explored the idea of using a four quadrant multiplier - specifically the AD734 (despite it's ridiculous price) to control the amplitude of a Wein Bridge. LTspice suggests that it would work rather well. The AD734 generates harmonics at around the -70dB level but when used as a gain correction mechanism the correction signal should be 60dB below the output, so the consequent harmonic level in the output should be better than 130dB below the fundamental.

I've also explored the idea of using a second - in quadrature - feedback loop to correct the frequency of the oscillation. That worked quite well in LTSpice, after some consultation with my friend in Scotland. Getting the quadrature signal required a somewhat messy phase shift network

You should be able to build an instant-start version of the circuit, but the jitter is never going to be as low as you can get with a faster oscillator running non-stop.

What was the amplitude variation and noise after filtering? I needed at least 96dB below signal and a few stages of filtering could actually generate more noise than it removes.

Yes, it took several re-designs to get 96dB, but I did eventually achieve it. Even potentiometers are too noisy for level control at those sorts of S/N ratios, so well-smoothed DC control of the oscillator amplitude was the only way. The gain-setting pot on the detector was too noisy for the most sensitive measurements, so it had to be left at one end of its travel and the oscillator output used to control the signal level instead.

It did mean that I found a completely unexpected extra use for the intermodulation meter: it could be used to test potentiometers and variable resistors for self-generated noise.

I was sort-of shocked when I came across a project with a local oscillator for a spectrum/network analyzer where the designer had used a DDS to provide the reference frequency for a PLL. The DDS was used just to cover the interval between two successive steps of the PLL. Quite over the top, I'd say.

Then again, frequency synthesizers with small steps can get complicated.

Jeroen Belleman the PLL.

Yup. The faster you can run the DDS/DAC the better (all else being equal, of course).

I put together something of this sort in an ICE40 FPGA, with a serialized bank of DDS phase accumulators to generate sine and ramp functions in one pipeline, a controllable-gain-per-input mixer in another, and an FM-stereo multiplex encoder in a third. I didn't have enough FPGA space to do a full 16-bit sinewave lookup, so that part of the pipeline uses a coarser lookup with linear interpolation - as I recall I got about 15.5 ENOB out of it. Most else is being done with

The output of the final mixer stage goes out serially to an audio DAC running at about 350 kilosamples/second. It produces a very nice FM-composite signal.

The output of the mixer is also added to a base increment, and the result sent out serially (at the same 350 ksamples/second rate) to an Analog Devices RF DDS.

The result is a very nice 10.7 MHz IF signal that I can use to analyze and adjust FM tuner IF stages and discriminators and MPX decoders. By mixing it with a 100 MHz carrier I can also test the tuner front end.

The very best tuner I've tried it with isn't actually intended as such/ It's an old Racal-Dana modulation meter. This meter uses a low-distortion pulse count discriminator, and has no IF filtering to speak of (very wide bandwidth). I got 0.03% THD for a mono signal at

Quite a fun learning experiment (my first serious Verilog work).

I've put some more information, including circuit diagrams at:

Thank you for the link, Liz. Your website's always fun to visit - for both its electronics and its music history. (One of my goals is to improve my own website.) My momentary muse happens to be fiddling with a crystal radio. Jan suggested using a linear power supply for it. And now your excellent intermodulation meter documentation serendipitously shows a suitable schematic:

I'll probably use some sort of linear regulator from the bone pile in place of the transistors.

Danke,

The voltage on the smoothing capacitors is perilously close to the 30v maximum input voltage of most commonly-available voltage stabilisers, that was why I used a descrete component pre-stabiliser instead of a chip.

Most stabilisers have rejection factor of about 60dB, so interaction between circuits on the same power supply rails isn't usually a problem, but I was trying to measure interactions 90dB down, so each section had its own stabiliser running off pre-stabilised lines that were common to all of them. Two steps of 60dB are much easier to achieve than one step of 120dB.

This equipment was intended for experimental work, rather than transmission line-up, so it had to go a bit further than usual. It was the result of 'mission drift' when I initially set out to disprove to a valve fanatic's assertion that second harmonic distortion was unimportant because it sounded "musical". (This is actually almost true for a single pure tone, but not for complex music waveforms.)

I was looking for an easy way to measure intermodulation distortion when I came across this idea in an article by Thomas Roddam in Wireless World in the 1950s. I don't think he could have actually built one or he would have discovered that his proposed iron-cored filter components would have caused more distortion than they removed. I think the original development of this method was done in the 1930s for the American film industry, where a notch filter for THD would not have worked because of the wow and flutter of the film sound track.

I do have a Marconi THD meter, which is basically a sharply tuned filter that can be swept through the audio band using the superheterodyne principle. As you say, it involves a lot of maths if you are interested in each harmonic individually -- but it can be used 'the other way up' to notch out the fundamental and measure all the other rubbish as a single reading. Taking very low THD measurements by adjusting the notch is an extremely slow and tedious business, compared with just operating a couple of switches and reading a meter for the I/M method.

The great advantage of the THD method is that it gives one number which the Sales Department can wave under the noses of the opposition, whereas I/M measurements will differ according to the pairs of tones selected (especially in amplifiers with iron-cored output transformers).

That is true, it always amuses me to see audiophiles comparing recordings when their listening was done on a car radio. ...or comparing amplifiers and loudspeakers using recordings that are almost totally synthetic.

Thank you for sharing your thought process and your intermodmeter documentation:

Empirical data indicates a zener-emitter-follower's rejection's superior to a typical TL431:

Integrated circuit complexity arguably adds noise, both in the literal and the figurative, metaphorical sense.

Danke,