Yamaha synth-expression circuit

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Hi all, first post here.
Please look at this picture

An externally hosted image should be here but it was not working when we last tested it.


It comes from a Yamaha polysynth; I'm studying this circuit, namely the PH1 and PH2 sections and I'd like to know :

1.Is it a kind of a loudness correction circuit?

2.How the two serial filters interact with the photo resistor?

3.Is that embedding able to make the filters change their slope according to the frequency of the fundamental? This is the main question: on this Yamaha polysynth , if you dropped the pitch down to very low pitches with a ribbon controller the sound became more "filtered" in a continuous way...someone told me that there may be a slope variation from 3dB to 12dB shelf and this may be related with the action of the Vactrol itself which controls the pedal too...
I wish for a simple yet precise explaination of how this circuit works, I'd like to try to build it when my skills will grow .
Thx for your care.
 
Excuse me, this topic relates to a musical instrument no doubt but may be a general question about amplification, I'm not really skilled in electronics. Moderators, is this the right place to have any explaination about the questions I put in my first post?
Thx for any reply
 
Hi Al, thx for your reply

The picture is a detail of a huge picture you can find

here , then on the right of this picture you can find the part I'm looking into just before the physical outputs

On my attachment the black arrows on the left are the audio signal channels going into the PH1 and PH2 filters network: the filters should act like a bass booster on the low pitches that acts with a psychoacoustic effect by reducing the trebles when the pitch goes under 700Hz with increasing force as the pitch becomes lower; note that the more the waveform is filtered through the master filters (HPF feeding a LPF on the synthesis path before the PRA-PH1 and PH2 that I'm showing here) , the more the sound seems "filtered" or damped at low registers, generally under 100Hz.
Thx for your help!
M
 
Ex-Moderator
Joined 2002
Ok a quick glance seems to back up your opinion. The buffer at the top drives the LEDs in the volume circuit, acting as an inverter, so as the pedal is pressed, the LEDs get dimmer.

Both sections of the filter do the same thing, so we'll just take the top as an example.

The resistor and filter network below the LDR act to change the frequency response as you guessed. However, as the LED gets brighter, the resistance of the LDR drops, so more signal passes though it, and less passes through the filter network, so decreasing its influence. Does that make sense?
 
Al
Seems correct when applying the pedal to the sound ( mind that PH3 section is just linked to the "wah-wah" circuit so that's it ), anyway , when playing the instrument I noticed a thing: even if you don't switch the pedal control ON the filters seem to act as they're tied to something , looks like the fundamental frequency because if I bend the frequency down to very low frequencies the filters of PH1-2 section react ; now the question is : while there is no real connection between any voice card ( vco ) and loudness filters, how can these filters change their cutoff or, better , their slope when you get the low pitch ? Does the photoresistor have any implication with this?
Thx for replying!
 
I've got some words from a guy

The two 68K resistors and their associated capacitors make a 2nd order lowpass filter. Since it is a passive RC filter, the frequency response will be very low-Q and gentle. It is basically two lowly first-order passive filters in series.
So the first filter -3 dB frequency, assuming that the 0.33 is microfarads, would be 1.0 / (2.0 * PI * 68.0 E3 * 0.33 E-6) = 7.09 Hz. The 2.2 K resistor in the first filter would limit the maximum high freq attenuation of the first filter. The max shelf attenuation should be reached somewhere around 220 Hz, and the max high-freq attenuation of the first filter would be in the ballpark of -30 dB.

The second filter -3 dB FC would be about 709 Hz.

and also

Max, here is another aspect of the opto attenuator I missed. Passive circuits don't have clearly-defined in and out points like active circuits or dsp stuff, so they can have sublties.

Assuming the output of the opto attenuator is feeding a pretty high-impedance buffer farther down the line (which would be typical), the CDS cell also makes a first-order lowpass filter against the 0.0033 uf capacitor.

Sometimes opto CDS cells go down into the low hundreds of ohms. It might be useful for you to eventually multimeter-measure the CDS cell with the pedal full down, if you can reach probes to the right point without tearing the synth apart.

Anyway, if the full-down resistance was 1 KOhm, the 6 dB per octave FC would be 48.2 KHz. When the resistance rises to 10 KOhm, the FC would be 4.8 KHz. The FC would continue to fall along with the amplitude. When the CDS cell resistance rises to 68K matching the other network resistors, the FC would be 709 Hz. As the CDS cell rises well above the 68K value, it would start approaching the 6 dB per octave slope descending from about -30 dB at 709 Hz, with the response gradually rising to -3 dB at about 7 Hz.

Or something in that ballpark. Pretty fancy behavior for 6 parts.

Can you explain me in a simplier way this "fancy behaviour" he's talking about?

Mostly my question is : Why if I bend the pitch down I get a sort of filtering at low pitch with a continuous variation ( filter opened towards mid pitch, closed towards low pitch ) ? If the expression filters are passive filters where should I look for this kind of behaviour??? Master filters or the VCO themselves ( could they loose harmonics at low pitch? )

BTW I have a short mp3 file splitted in two parts, the first gets an organ-like preset ( basically one sawtooth with mid filtering ) first low pitchm, then coupled with high pitch, then only high pitch bent down to zero, the second features the same preset with increased resonance in order to enhance this behaviour...how can I put it there?
M
 
Ex-Moderator
Joined 2002
omissis said:
Can you explain me in a simplier way this "fancy behaviour" he's talking about?

He has taken his analysis a bit further than I did, (he did the maths!), but as far as I see it he is just commenting on the elegant circuit design that does a lot with few components.

I suspect the behaviour you're hearing is actually in the synth, from what section I wouldn't like to guess. Why are you interested? Do you want to try and copy it or is it something you want to get rid of?

BTW I have a short mp3 file splitted in two parts...how can I put it there?

We don't accept audio files here, but we can take zip archives. ;)
 
Why I'm interested in that? Because I'm learning electronics and I found those features some nifty ones which give a contribute to the sound and seem to have been forgotten by the actual design/programming with concerns to Virtual Analog...and yes, I want to copy it (one day)....for the moment I'm helping a friend restoring his Yamaha CS80; since these features I didn't find ever on a synthesizer and since I'm still not so skilled about theoretical electronics , I'd like to know something more 'bout this captivating machine...:hbeat:
 
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