[time-nuts] Noise of digital frequency circuits (was: Programmable clock for BFO use....noise)
k8yumdoober at gmail.com
Mon Sep 17 01:37:43 EDT 2018
The act of squaring up the waveform alone might not do much harm, depending
on the extent
to which the phase noise on said waveform has already been filtered off.
But it's mainly when
the signal gets divided down by large ratios that the difference would
become really noticeable.
For example, take the case of 10 MHz starting frequency; the phase noise
several MHz out
is likely to be nil. But divide the 10 MHz down to, say, 1 Hz; then
there's likely to be quite a
lot of phase noise within "folding range" of many Nyquist bands about 1 Hz.
This, again, is why I wonder so much about our efforts in re-synthesizing
higher frequencies from
the 1PPS from GPS receivers. I don't know much the architecture of GPS
receivers, but it seems
to me it would sure be nice if there were some convenient way to extract a
clean signal at the
chipping rate, for use in generating standard reference frequencies.
On Sun, Sep 16, 2018 at 9:15 PM, Bob kb8tq <kb8tq at n1k.org> wrote:
> It’s pretty easy to demonstrate that squaring up a sine wave, even with
> fairly simple
> circuits does not create crazy phase noise issues. People have been doing
> it successfully
> for a lot of years. In general faster saturated logic produces lower noise
> floors than slower
> > On Sep 16, 2018, at 4:33 PM, Dana Whitlow <k8yumdoober at gmail.com> wrote:
> > I'd been thinking, in an admittedly non-rigorous sort of way, about this
> > matter for some years.
> > As I see it, it is certainly true that the phase of an oscillator's
> > is a continuous funciton
> > of time. It could be described as a continuous ramp, whose slope
> > corresponds to the frequency,
> > and with a little bit of non-flat random noise superimposed on it.
> > Now if you square up the waveform and do digital things with it (such as
> > freq dividing, digital
> > phase detection, etc), you are really only glimpsing the phase noise at
> > transition times, and
> > are blind in between. Thus the very process amounts to sampling the
> > noise waveform
> > with a sampling phase detector. This view suggests that all the phase
> > noise power is aliased
> > and folded back into the band ranging from DC to Fsamp / 2, where Fsamp
> > the frequency
> > of the waveform after frequency division. This is why the time domain
> > jitter of the oscillator's
> > waveform is unchanged by "perfect" frequency division (or
> > It is why I wonder about the wisdom of doing phase comparison at
> > unnecessarily low frequency-
> > all that noise would seem to be scrunched down into a bandwidth of half
> > comparison frequency.
> > Does this explanation help, and how does it sit with those of you who
> > more expertise
> > than I?
> > Dana
> > On Sun, Sep 16, 2018 at 4:06 PM, Attila Kinali <attila at kinali.ch> wrote:
> >> Moin,
> >> On Sat, 15 Sep 2018 08:38:55 -0700
> >> "Richard (Rick) Karlquist" <richard at karlquist.com> wrote:
> >>> On 9/15/2018 3:26 AM, Attila Kinali wrote:
> >>>> possible logic family for the task. Otherwise the harmonics of the
> >>>> switching of the FF will down-mix high frequency white noise down
> >>>> to the signal band (this is the reason for the 10*log(N) noise scaling
> >>>> of digital divider that Egan and Calosso/Rubiola and a few
> >>>> mentioned).
> >>> Wow, I never knew this in 45 years of designing synthesizers!
> >>> I do remember that some of the frequency counter engineers at HP
> >>> talked about noise aliasing. I think this is another way of
> >>> describing the same problem.
> >> Yes. This effect has been known for a few decades at least.
> >> What kind of puzzles me is, that I have not seen a mathematically
> >> sound explanation of it, so far. People talk of aliasing and sampling,
> >> but do not describe where the sampling happens in the first place.
> >> After all, it's a time-continuous system and as such, there is no
> >> sampling. One could look at it as a (sub-harmonic) mixing system,
> >> but even that analogy falls short, as there is no second input.
> >> It also fails at describing why there is not infinite energy being
> >> down-mixed, as the resulting harmonic sum does not converge.
> >> If someone knows of a description that goes beyond handwavy arguments,
> >> I would very much appreciate hearing of them.
> >> The only way to explain the effect in a rigorous way, that I could
> >> figure out, is to apply Hajimiri and Lee's Impulse Sensitivity
> >> and adapt from the oscillators they discribed to general periodic
> >> (The step, as one can guess, is small, but hic sunt dracones)
> >> Doing this, it becomes obvious that the down-mixing is an inherent
> >> property of all systems that use or generate non-sinusoidal waveforms.
> >> It is this ISF that is the source of the down-mixing/aliasing effect,
> >> as it has a periodic waveform of sharp spikes.
> >> As the ISF is probably (this is my intuition and I have, unfortunately,
> >> no proof of this) related to the derivative of the produced output
> >> waveform,
> >> it becomes important to limit the slew rate of the output, to introduce
> >> a second pole in the ISF and thus limit the number of harmonics.
> >> Yet, it is also important to keep the input slew rate high, in order to
> >> keep the width/height of the ISF pulses low.
> >> A partial discussion of this can be found in the paper I presented
> >> at IFCS earlier this year. Unfortunately, the write-up is not
> >> nice and I only realized after the deadline that I should have
> >> all written it using a different approach. Sorry for that.
> >> If something is not clear, do not hesitate to send me an email.
> >>> About 10 years ago, the frequency synthesizer chip vendors started
> >>> talking about a Figure of Merit (FOM) that predicted phase noise floor,
> >>> and it also included the 10 LOG N noise scaling. An application
> >>> engineer at ADI told me this was a characteristic of the sampling phase
> >>> detector that all these chips used. But I always wondered if the
> >>> frequency divider could come into play. The way FOM is defined,
> >>> it doesn't distinguish between phase detector and divider noise.
> >> The 10*log(N) also applies to the phase detector in PLL chips,
> >> where N becomes the ratio of the phase detector bandwidth divided
> >> by the phase detector input frequency.
> >> Given that the phase noise is dominated by the input source' phase
> >> noise, there will be no appreciatable difference in whether the
> >> down-mixing happens in the divider or the phase detector, as long
> >> as the bandwidth of all components is the same. If the bandwidth
> >> is different, we get into something akin Collins' zero crossing
> >> detector where appropriately designed stages with different
> >> input bandwidths limit the energy that is down-mixed.
> >>> At Agilent, we used to make a lot of lab demos using a Centellax
> >>> (now Microsemi AKA Microchip) frequency divider that could divide by
> >>> number between 8 and 511 up to 10 GHz. It was absolutely fabulous for
> >>> dividing 10 GHz down to 2.5 GHz. But 20 LOG N quit working if I tried
> >>> to divide down to 50 MHz. Now you have explained it.
> >> Hmm? Are you implying those chips somehow were able to give
> >> a 20*log(N) phase noise behaviour? If so, do you know how
> >> they achieved such a feat?
> >>>> If you divide by something that is not a power of 2, then it is
> >> important
> >>>> that each stage produces an output waveform with a 50% duty cycle.
> >> Otherwise
> >>>> flicker noise which has been up-mixed by a previous stage, will be
> >> down-mixed
> >>>> into the signal band, increasing the close-in phase-noise.
> >>> Wow, another thing I never knew.
> >> I do not think that anyone was aware of this. A least I do not remember
> >> seeing this being mentioned in any of the papers I have read. I, myself,
> >> stumbled over it by accident. I was trying to design a sine-to-square
> >> wave converter and wanted to understand what happend to the noise.
> >> Especially the AM to PM conversion that a few people here have mentioned
> >> a few times. I was looking at Claudio's measurement [4, page 28] and,
> >> after applying Hajimir and Lee's ISF, I could (mathematically) explain
> >> everything but what Enrico so nicely labled as "bump". None of the
> >> explanations that I exchanged with Enrico, Claudio, Magnus and a few
> >> other people made sense with the complete data. An external influence
> >> didn't make sense as the flicker noise went from a straight ~6dB/oct
> >> to a straight ~3db/oct line below 25MHz. This hunch got stronger when
> >> Claudio shared the complete circuit they used with me(see figure 3 in
> >> The feedback circuit, which stabilizes duty cycle, has a -3dB frequency
> >> of 0.28Hz, which is exactly the frequency where the bump is. And below
> >> it, the flicker noise behavior seems to go back to approximately
> >> For a complete explanation, see my paper section 5.D "Scaling in a
> >> Multi-Stage Sine-to-Square Converter."
> >>> The conventional wisdom was to
> >>> divide by any number (even or odd) and then follow that divider
> >>> with a divide by 2 flip flop to get 50%. Now, that is in question.
> >>> The now correct answer is to us a variable modulus prescaler to
> >>> divide by P and P+1, controlled by a toggle flip flop to make
> >>> half the divisions at P and half at P+1.
> >> I don't think the modulus prescaler is a good approach.
> >> It will help reduce flicker noise, at the price of incrased
> >> white noise, as the two division values will generate two
> >> frequency spikes in the ISF that are close to each other.
> >> There is probably some residual even harmonic content due to
> >> the switching betwen the two scaler values, which will increase
> >> flicker noise, not as much as having non-50% duty cycle, but still.
> >> The right way to do it is to use both edges in case of odd division
> >> factors (as some of the divider circuits by Linear/Analog seem to do).
> >> Alternatively generate a ramp/sine output, ie use a Λ-divider
> >> or a DDS, as both have much lower harmonics content in the ISF
> >> and thus do not suffer from the down-mixing as much. If a square
> >> waveform is required afterwards, a square-to-sine converter with
> >> approriate bandwidth for the output frequency will solve that.
> >> Attila Kinali
> >>  "A General Theory of Phase Noise in Electrical Oscillators,"
> >> by Hajimir and Lee, 1998
> >>  "A Physical Sine-to-Square Converter Noise Model,"
> >> by Kinali, 2018
> >>  "The Design of Low Jitter Hard Limiters," by Collins, 1996
> >>  http://rubiola.org/pdf-slides/2016T-EFTF--Noise-in-digital-
> >> electronics.pdf
> >> --
> >> <JaberWorky> The bad part of Zurich is where the degenerates
> >> throw DARK chocolate at you.
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