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cebec

Joined: Apr 19, 2004 Posts: 1097 Location: Virginia
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G2 patch files: 31
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Posted: Mon Dec 12, 2005 2:17 pm Post subject:
Mixer Math |
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I'm continually impressed by Rob's workshops, patches, and contributions. I often dissect and cannibalize his patches for the terrific 'building blocks' included within them.
However, there're at least two areas of many of these building blocks that I have difficulty understanding; the math performed with the mixer modules and the feedback paths.
For example, Rob's latest Matrix Synth has a modified NM Classic filter which uses a few mixers to add a DIY resonance control and probably to tune the filter differently. It sounds terrific!
If I could understand the basic rules and principles for using such feedback paths and performing math with mixers I could learn to design or at least tweak some DIY circuits.
Can Rob, Tim, et al. provide any insight into this? Where do I begin?
Thanks! |
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Tim Kleinert
Joined: Mar 12, 2004 Posts: 1148 Location: Zürich, Switzerland
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G2 patch files: 236
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Posted: Mon Dec 12, 2005 4:07 pm Post subject:
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Cebec,
I don't have the patch you are referring to so I can't give specific answers. Maybe if you could post a link or a specific question... |
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cebec

Joined: Apr 19, 2004 Posts: 1097 Location: Virginia
Audio files: 3
G2 patch files: 31
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Tim Kleinert
Joined: Mar 12, 2004 Posts: 1148 Location: Zürich, Switzerland
Audio files: 7
G2 patch files: 236
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Tim Kleinert
Joined: Mar 12, 2004 Posts: 1148 Location: Zürich, Switzerland
Audio files: 7
G2 patch files: 236
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Posted: Mon Dec 12, 2005 4:34 pm Post subject:
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OK, I checked it out. As far as I can see:
The first mixer in the blue filter section serves to bandlimit and boost the signal at the same time. (Boosting happens because the sum of the input levels is bigger than unity gain. In this case this boosting is desired because the signal has to be sufficiently boosted anyway within a filter feedback path i n order for resonance to occur. This mixer serves both purposes neatly.)
The second mixer inverts the signal and adds some of the original signal before sending it back into the loop. The inversion is needed for the resonance to work properly with a 24dB lowpass filter, and the "original signal injection" serves to combat the passband loss that usually occurs at high resonance settings. This makes the filter sound "bassier" even with alot of resonance. |
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cebec

Joined: Apr 19, 2004 Posts: 1097 Location: Virginia
Audio files: 3
G2 patch files: 31
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Posted: Mon Dec 12, 2005 4:42 pm Post subject:
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Thanks, Tim. I'm just processing your analysis right now.
Are these design choices based in the convergence of both DSP and subtractive synthesis? Or is it something else? |
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Tim Kleinert
Joined: Mar 12, 2004 Posts: 1148 Location: Zürich, Switzerland
Audio files: 7
G2 patch files: 236
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Posted: Tue Dec 13, 2005 7:03 am Post subject:
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cebec wrote: | Thanks, Tim. I'm just processing your analysis right now.
Are these design choices based in the convergence of both DSP and subtractive synthesis? Or is it something else? |
Well, the bandlimiting is done for reasons stated above. And this is in fact analogous to analog circuitry (no pun intended), as analog circuits have to be bandlimited too, in order not to start self-oscillating at radio frequencies (and act as radio receivers).
The "extra signal injection to combat passband loss" is purely an esthetic design choice. An analog lowpass filter exhibits the same amount of passband loss when driven into self oscillation, and IIRC there are some analog designs that feature this extra injection circuit too.
So, these concepts do not only apply to digital DSP, but also to analog.
However -the "mixer feedback bandlimiter" is a G2-specific trick, as the output of the mixer is always one sample behind of the input. (I wouldn't try to feedback your old Mackie mixer ...) |
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Rob

Joined: Mar 29, 2004 Posts: 580 Location: The Hague/Netherlands/EC
G2 patch files: 109
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Posted: Sun Dec 18, 2005 4:37 pm Post subject:
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Allow me to explain the how and why. I was not really satisfied with the amplitude drop of the Classic filter when the resonance is raised. The Classic filter has a little bit of compression built in, which is absent on the other filters, but the drop in amplitude does not really make use of this. I wanted to improve on these matters.
The old trick to compensate for a drop in amplitude in an analog ladder filter (or digital emulation) is to add some of the input signal just before the internal mixer knob that controls the resonance. But as this signal path is a 'module internal' path, the only solution is to create an extra external signal path for resonance, and not use the resonance knob on the filter module anymore. As the amplitude is dropped by about -12dB for a high resonance setting it means that quite a lot of the input signal must be added. Plus 9 dB seems just about right to both compensate for the overall signal drop and make use of the compressive effect in the filter to get a slightly saturated resonance peak. In the resonance feedback path the signal must also be amplified and slightly filtered to prevent high frequency oscillation if the filter is fully opened.
All in all everything can be done with just three extra mixer modules. The first mixer module acts as both the new signal input (on the chain input) and the resonance control knob. Then comes the filter module itself, and this module also has the filtered output signal on its output. The filtered output signal is then amplified and slightly filtered in the next mixer to create a properly boosted resonance feedback signal. The last mixer receives the input signal on two inputs to add the extra 9 dB for a high resonance setting, and inverts the boosted resonance feedback signal. The output of the last mixer is then routed back to the first mixer.
So, if the resonance control is closed the input signal will just fall through the first mixer into the filter module and there is the standard non-resonant filtering effect.
If the new resonance knob is opened it will not only feed the resonance signal back into the filter module input, but also add a 9 dB of extra input signal. Now you can clearly hear the effect of the compressive saturation in the Classic filter module, which produces extra odd harmonics.
It is possible to add even more saturation to get a distortive filter sound. This can be done by e.g. adding an extra non-linear gain cell in the resonance feedback path. An example is in the second attachment where some extra even harmonic distortion is added. Note that as this is an asymmetrical type of distortion (a positive signal is boosted while a negative signal is compressed) only a little bit must be used to avoid the resonance feedback signal to clip to the positive headroom limit.
Description: |
Classic ladder filter with compensation for the amplitude drop at higher resonance settings |
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Filename: |
Robs ladder filter.pch2 |
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1.52 KB |
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1603 Time(s) |
Description: |
Classic ladder filter with compensation for the amplitude drop at higher resonance settings and some extra even harmonic distortion |
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 Download |
Filename: |
Robs ladder filter with even harmonic distortion.pch2 |
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1.5 KB |
Downloaded: |
1660 Time(s) |
Last edited by Rob on Sun Dec 18, 2005 5:04 pm; edited 1 time in total |
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Rob

Joined: Mar 29, 2004 Posts: 580 Location: The Hague/Netherlands/EC
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cebec

Joined: Apr 19, 2004 Posts: 1097 Location: Virginia
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Posted: Sun Dec 18, 2005 8:01 pm Post subject:
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Thank you, Rob. This is all very helpful! |
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Afro88

Joined: Jun 20, 2004 Posts: 701 Location: Brisbane, Australia
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Posted: Mon Dec 19, 2005 5:48 pm Post subject:
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Thanks very much Rob! |
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