[sdiy] Frequency follower idea. Question about FM.

Giorgio dancemachine at gmail.com
Thu Sep 9 18:37:12 CEST 2010


good stuff. you might want to look at the harvestman hertz donut if you haven't already. it has some very interesting and musical oscillator sync/tracking behaviors

sent from phone

On Sep 9, 2010, at 12:12 PM, cheater cheater <cheater00 at gmail.com> wrote:

> Hi guys,
> I was wondering recently about syncing VCOs to other oscillators, and
> I came up with a framework idea of how this could be done in a new and
> I think interesting way. I wonder what you guys think about it.
> 
> One of the problems is: you want an analogue LFO that is tempo-synced,
> but you want e.g. a sinewave, or a triangle wave, and very importantly
> it must reach the amplitude maximum. If you just coarsely set the
> frequency of the VCO and then sync it to sequencer pulses, then you
> lose the amplitude (the typical DCO problem). If on the other hand you
> don't want to lose the amplitude (because, for example, the LFO
> modulates the pitch and it's important that it reaches a certain
> interval), then the only other way for the VCO to catch up is for its
> pitch to be raised temporarily. It's similar to how beatmatching works
> with vinyl. Of course, this can work well for audio-rate VCOs as well.
> 
> So here's the idea with how vinyl works vs how VCOs work. We have two
> VCOs vs two adapters with vinyl records on them.
> 
> 1. unsynced VCOs - two records are playing with separate tempo. There
> is no sync.
> 
> 2. Hardsynced VCOs: every time VCO1 reaches its sync point, the VCO2
> is also reset towards its own sync point. This can be seen as speeding
> up VCO2 to a very high speed of oscillation until it reaches its sync
> point. The analogy here would be that when vinyl 1 reaches a kick,
> vinyl 2 is sped up to nearly-infinite speed until it reaches the kick.
> Since the speed is infinite, the operation takes an infinitesimal
> amount of time, and in the next measure vinyl 1 and vinyl 2 start out
> perfectly in phase. Because they are probably set to different tempos
> (the probability of having the same tempos is 0), they will quickly
> drift out of sync.
> 
> 3. Beatmatching: this is how records are synced: vinyl 1 is playing at
> a certain rate, vinyl 2 is playing at another rate. First the two are
> synced together. Then the dj checks if vinyl 2 is slower or faster and
> adjusts the rate of deck 2 to be higher or lower. At the same time he
> either pushes vinyl 2 forward or taps it with the finger to slow it
> down.
> 
> This could look like this with VCOs: You sync the two VCOs together.
> You have access to the phases of both VCOs and check how quickly they
> fall apart. The derivative of their difference is the difference in
> frequency that needs to be added to VCO2. Then, the oscillators are in
> frequency sync (let's call this part we have just accomplished
> "frequency calibration"). However, after this happens, the oscillator
> needs to be 'pushed forward'. We can do this by injecting some phase
> to the VCO. However, so that there is no click*, we can limit the rate
> at which the VCO is being "pushed forward". We can do this by
> frequency-modulating the VCO one time with a windowing function to
> speed it up, or inverted windowing function to slow it down (let's
> call that part "phase calibration"). During the phase calibration, the
> frequency calibration should be somehow disabled. However, in a
> running process, it would be undesirable to have a state-machine, so
> maybe some form of continuous process could be invented from this.
> 
> I have below included the mathematics behind this technique. One
> question is how to realize a fairly smooth windowing function.
> 
> *I believe if we have a carrier at frequency f_1 and it is modulated
> by something that has frequency content (much) lower than f_1, then
> you should not get any new audio-rate harmonics that are not "part of
> f_1", should you? How does this work exactly? Is there a special rule
> that says that your modulator needs to be band-limited to a certain
> frequency dependent on f_1? I understand that this question is not
> stated 100% properly, but any hints in this region are good. Basically
> I have an audio-rate carrier, and I want to frequency-modulate it so
> that no clicks/added harmonics are perceived in the output.
> 
> Also on a similar note, the MF-107 FreqBox: Does anyone know how this
> thing functions? Does it have a PLL inside? I assume it has a PLL and
> a VCO, and the PLL can sync the VCO, while the original signal can FM
> the VCO. An env follower can FM and AM the VCO.
> 
> Here are the maths for the calibration:
> 
> p_1, p_2
> phases of oscillators 1 and 2
> 
> r = p_1 - p_2
> Phase difference between oscillators 1 and 2.
> 
> dr/dt
> the rate of change of the phase difference.
> 
> If for example r is negative, and keeps on falling (we are
> "unwrapping" r, instead of counting it modulo 2pi) then raising the
> pitch of vco2 a bit will make the difference r drop a little less.
> 
> Therefore, we need to heighten the frequency by -h*dr/dt, where h is
> some constant:
> 
> f_2 |-> f_2 + (-h) * dr/dt
> 
> After this happens, dr/dt ~= 0 (is nearly equal to zero). Therefore r
> is constant. We need to inject the amount r of voltage into oscillator
> 2's accumulator:
> 
> r = p_1 - p_2
> p_2 = p_1 - r
> 
> if we want p_2 to be equal to p_1, we need to add something to it:
> 
> p_2 + x = p_1
> p_1 - r + x = p_1
> x = r
> 
> Therefore we need to add r to p_2 :)
> 
> To add r to p_2 we can do a frequency modulation by a window function w(t).
> 
> The (unwrapped) phase of vco2 called p_2 is the integral of its frequency f_2:
> 
> p_2(t_now) = b Int_0^{t_now} f_2(t) dt + p_2(0)
> 
> That is, the phase at time t=t_now is the integral from time t=0 to
> t=t_now of the frequency control voltage f_2(t) over the time-variable
> "t", multiplied by an amplitude "b" (which depends on the design of
> the VCO), and all that added to the phase it had at t=0.
> 
> similarly, for vco1:
> 
> p_1(t_now) = b Int_0^{t_now} f_1(t) dt + p_1(0).
> 
> We assume that f_1=f_2=const starting at some point we'll start
> calling t=1. The number 1 is chosen arbitrarily. It might mean t=1
> second. At t=1, the accumulators had phases p_1(1) and p_2(1) and the
> VCOs were at the same frequencies, meaning the phase rises at the same
> speed. We rebase our integral and with those constraints we can start
> counting it from t=1.
> 
> This means that
> p_1(t_now) = b Int_1^{t_now} f_1(t) dt + p_1(1)
> and
> p_2(t_now) = b Int_1^{t_now} f_1(t) dt + p_2(1)
> 
> therefore for t_now > 1, p_1(t_now) - p_2(t_now = p_1(1) - p_2(1).
> This difference is our, now constant, value r.
> 
> We need to add r to p_2 somehow. This means that starting with t_now >
> T, we want this to be the case:
> 
> p_2(t_now) = b Int_1^{t_now} f_2(t) dt + r + p_2(1)
> 
> The only parameter we can manipulate is f_2(t). If we do not
> manipulate it, after t=1 it will stay equal to f_1(t) and therefore
> constant.
> 
> So we can do this:
> 
> p_2(t_now) = b Int_1^{t_now} f_2(t) + k*w(t) dt + p_2(1)
> 
> = b Int_1^{t_now} f_2(t) dt + b Int_1^{t_now} k*w(t) dt + p_2(1)
> 
> But what we want is p_2(t_now) = b Int_1^{t_now} f_2(t) dt + r + p_2(1)
> 
> Therefore, we need to choose w(t) such that:
> b Int_1^{t_now} k*w(t) dt = r
> 
> We know b (it comes from the design of our oscillator), we know r, we
> need to find k and w(t).
> 
> We need our function w to be such that its integral is 1 when counted
> from point t_0>1 where we begin emitting it, until T where we will
> have stopped emitting it. We also want it to not add any phase
> afterwards, that means, for t>T w(t)=0. And we don't want it to add
> any frequency before we start emitting it, that means for t<t_0,
> w(t)=0. This means that:
> 
> Int_{t_0}^T w(t) dt = 1
> 
> for u < t_0 < T < v: Int_u^v w(t) dt = 1
> 
> So we have chosen our window function, therefore we have w(t).
> 
> Our equation looks like this:
> 
> b Int_1^{t_now} k*w(t) dt = r
> 
> b*k Int_1^{t_now} w(t) dt = r
> 
> Because 1 < t_0 and t_now > T, we can say:
> 
> b*k * 1 = r
> 
> Therefore k = r/b.
> 
> This means we need to inject our windowing function once, while its
> amplitude is multiplied by the number r/b.
> 
> I believe it is possible to extend this technique to a continuous time
> control system. The technique above will only work for when VCO1 has a
> constant frequency. A continuous control system could work for the
> case when VCO1 is of changing frequency.
> 
> This technique could sound fairly differently depending on what
> windowing function is chosen.
> 
> A continuous technique could work like this:
> 
> We define two areas in which p_2(t) can be:
> (A) it's either in in the area p_2(t) \in (p_1(t)-pi, p_1(t) ) <mod 2pi>
> 
> (B) or it's in the area p_2(t) \in (p_1(t), p_1(t) + pi) <mod 2pi>
> 
> The cases p_2(t) = p_1(t) and p_2(t) = p_1(t)+pi <mod 2pi> are
> excluded for clarity, since the probability they will happen is 0.
> 
> There is a comparator that outputs 1 for case (A) and -1 for case (B).
> Let's call it L(t).
> 
> This means, for example, if VCO2 is just a tiny amount behind VCO1,
> L(t) will be +1, and if it's just in front of VCO1, L(t) will be -1.
> 
> Assume we have case (A). Every now and then, while L(t) is positive,
> we feed a small amount of charge, say K, into the phase accumulator
> p_2(t) to "catch up" and perform phase calibration. As soon as we
> overtake VCO1, L(t) will become negative, and K will also become
> negative, therefore "slowing down" p_2(t). This means that our p_2(t)
> will be jumping back and forth from being just under to just above
> p_1(t).
> 
> Of course a continuous version of summation is an integration. So a
> way to do the above is to have an current following L(t) and add that
> current, call it M(t), to p_2(t). (I am, of course, not good enough
> with electronics to work out the specifics of that).
> 
> However, we still need to make sure that the change only happens
> gradually: starts up slow, speeds up, and ends slow. Therefore before
> we inject M(t) into p_2(t), we pull it through a sine shaper. The
> shaper should somehow auto-adjust so that the electric charge
> injection becomes slowest as p_2 approaches p_1. I am not sure what
> the specifics could be of this. Does anyone have any ideas?
> 
> To prevent oscillation of the system when p_2(t) ~= p_1(t) and when
> p_2(t) ~= p_1(t) - pi <mod 2pi>, we put a low-pass filter of some
> frequency F on M(t) so that it does not "zipper" along with L(t). In
> the latter case the system should be knocked out of place into either
> state domain (A) or state domain (B) by the fact that frequencies are
> different.
> 
> Of course, of f_1 > f_2, then we would need to keep on emitting M(t)
> indefinitely, because p_2(t) would keep on lagging behind.
> 
> However, the closer we get to the state p_1(t) ~= p_2(t), the easier
> it will be for us to start marking up the frequency. This is because
> the rate of change of r will be lower and lower. This means that we
> have a system that is now under control and we can slowly start doing
> other things with it. We look at L(t) statistically. If f_2 < f_1,
> then L(t) will be mostly positive. If f_2 > f_1, then L(t) will be
> mostly negative. The closer f_2 is to f_1, the more even the
> distribution of L(t) will be across the values +1 and -1. We can take
> the low-pass-filtered M(t) and inject it into some form of
> sample-holding capacitor. The charge of that capacitor will be
> mirrored into some form of positive voltage which raises or lowers the
> frequency of VCO2. As f_2 -> f_1, M(t) will tend to 0, and the
> frequency of f_2 will stabilize.
> 
> This technique could work better than the previous one for the
> situation where f_1 changes. However the faster f_1 changes, the
> higher the frequency of the lowpass filter on M(t) should be, so that
> f_2 has enough resource to catch up. Therefore: F(t) ~ f_1'(t) (note
> the derivative sign)
> 
> This fourth approach can be analogous to the DJ catching up with two
> hands on the record adapter 2, trying to run his hands so quickly that
> adapter 2 is in sync with adapter 1; then, a second person starts
> correcting the pitch, during which the dj slows down his hands and
> finally lets go of the record what so ever.
> 
> Of course both techniques I described above can also be enhanced to
> allow keeping the two oscillators at a certain desired difference of
> phases r = r_0. This could prove sonically interesting.
> 
> You can assume that the second technique will not work *ideally*, that
> is, there will be some sway in the value of r, even if f_1 is
> relatively constant. This will probably create a somewhat random
> phasing sound which is very pleasing to the ear.
> 
> If the low pass filtered M(t) is completely disconnected from the core
> of VCO2, L(t) can be used as a measure of how close together f_1 and
> f_2 are. You could do it this way: you check how symmetric L(t) is
> (over a short period of time) and the longer it is symmetric, the
> higher voltage you output from an integrator. On the other hand if
> L(t) stops being symmetric, the charge in the integrator is reduced.
> The integrator should have a low rise rate and a high fall rate. It
> can be used to amplitude-modulate VCO2, making VCO2 output only when
> VCO1 is near its frequency. This can be used for a resonator. Because
> of the relatively low parts count for a minimum implementation using a
> fixed frequency LPF for M(t) and a fixed frequency oscillator for
> VCO2, you could easily imagine banks of such oscillators, working as a
> very interesting resonator bank. This could be very musically
> pleasing, giving a sweeping tone from VCO1 a nice arpeggiated
> characteristic. You could even use some sort of fixed AD envelope to
> give the resonances a plucked sound.
> 
> The electronic implementation is left as an excercise for the reader.
> 
> Cheers,
> D.
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