SSL "Talkback" Listen Compressor

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Ok, so 470nF it is.

They must have specced it for when the recording eng. yells at the _talent_ in the rec.room:
"Tune up your guitar, you stupid stoned idiot you!"

the eng. hears:
"Yessss, ssssr.."

instead of:
"Yeffff, ffffr.."
 
Posted another revision. Realized I put C3 inline on the 48V input instead of one leg going to ground. C3 could probably be left out anyways but I'm going to keep it on I think. Dunno I have some nice 3u3 caps lying around.  OR should I just leave out C3?

Another anomaly between the board and the schematic is C16 which is actually another 10uF/35V cap instead of 100uF thats on the schematic. 100uF might not be a bad idea there though anyways, but I put 10uF in my schemo
 

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  • Talkback_Limiter_REV3.pdf
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geetar_king said:
Posted another revision. Realized I put C3 inline on the 48V input instead of one leg going to ground. C3 could probably be left out anyways but I'm going to keep it on I think. Dunno I have some nice 3u3 caps lying around.  OR should I just leave out C3?

Another anomaly between the board and the schematic is C16 which is actually another 10uF/35V cap instead of 100uF thats on the schematic. 100uF might not be a bad idea there though anyways, but I put 10uF in my schemo

Any News or developments in this project?

 
geetar_king said:
Yes the production boards actually have 470nF caps there for both talkback amps! Strange there is a discrepancy on the SSL schematic.

Probably not at all strange. AFAIK many of the original SSL schemos had 'little mistakes', not so bad that clients couldn't fix the units - but in order to make it seriously more difficult to clone them. Back in those days cloning was done by pros, meaning competitors...
Never trust a SSL schemo on it's own, either use your brain or an original module to verify (I normally choose the module  ;) )!

Michael
 
Yes Whoops! Working on a first proto right now. Working with Livingnote on getting a CnB board combined with the talkback board... Hopefully will have something assembled and powered up in the next few weeks.

edit: and by "Yes Whoops!" i don't mean "Yes Oops!" haha
 
geetar_king said:
Yes Whoops! Working on a first proto right now. Working with Livingnote on getting a CnB board combined with the talkback board... Hopefully will have something assembled and powered up in the next few weeks.

edit: and by "Yes Whoops!" i don't mean "Yes Oops!" haha

Great news,
nice to Know that.

wish all good for the next weeks
 
So, for the blend, I have made an attempt to modify Lucas' and Keith's work to allow for balanced output. Basically stuck a SSM2142 on the output to balance it.

Also added gain control for the 'wet' compressed signal.  The schematic attached is for dual mono. The talkback will hook up to the blend board as discussed previously in the thread. So a clean takeoff after the input gain stage and a 'wet' takeoff at the end of the talkback chain. 

Might have to play with some resistors to get it to fly, but do you guys think this blend schematic should work?  ???


(You need to be logged in to view the attachment)
 

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  • Talkback Limiter Blend_R1.pdf
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fucanay said:
This thread was great reading and then it just stopped. Any updates on this?

Matt

I know... I've been a little lazy lately.  Right now I have the blend and talkback boards populated and by sunday I will have everything hooked up and ready to test. I'll hopefully have it all together today. There might be a little tweaking work to do, but we'll see.
 
wired up a channel last night but didn't have much time to play around with it.  it's passing audio! phew!  still some work to do with the blend board.  man this thing really compresses though when it kicks in!  i put it on a snare track and i could almost here my neighbor's microwave coming through in the background  :p

anyone know what the best way is to set the bias trimpot for the FET?
 
thanks! hopefully get it tweaked over the next few days here. i'll get some pictures up sometime too

i guess setting the bias on the FET would be like adjusting the Q bias on the 1176 eh?

i'm using gustav's PSU board for supply and my 48V is actually about 58V for some reason.  will 58V before the 6K8 resistors hurt anything using phantom power or should I fix this? i guess the higher voltage means higher current.. maybe i'll tweak the PSU must have got a wrong resistor in there or something...
 
geetar_king said:
wired up a channel last night but didn't have much time to play around with it.  it's passing audio! phew!  still some work to do with the blend board.   man this thing really compresses though when it kicks in!  i put it on a snare track and i could almost here my neighbor's microwave coming through in the background  :p

anyone know what the best way is to set the bias trimpot for the FET?

On most FET type compressor/limiters, you first want to adjust the trimpot so there is no bias on the fet.

Looking on the schematic, you would remove jump 2 so that when you apply a test audio signal the side chain is not trying to compress.

apply a tone signal to the input (1kHZ will do), and monitor the output with a device that can read dB out. adjust the gain of the signal so it high enough to read on the output of U1B, but not high enough to distort.

slowly turn the bias trim pot until the output of U1B drops by 1dB. This may take a little effort as the bias circuit has a long time constant in this circuit. What this does, is turn the FET channel on so it's always conducting whether you are compressing or not. This reduces distortion.

return the jump 2 to it's place, remove the test signal and apply audio.

Compress and enjoy.
 
Got it biased like you said, and now it sounds great!  Talkback board doesn't need any tweaking as far as I can see right now.

The only problem i'm having is with the blend board. (Schematic posted below again so you dont have to go back and find it)

Basically the talkback board has two outputs, one for clean original signal and one for compressed signal.
The blend board has two inputs one for the clean original signal that passes in at unity gain, and the other input which I'm trying to set up for 20dB makeup gain for the compressed signal.

If I pass either output from the talkback through the 'clean' side of the blend board (ie no makeup gain pot-side) then everything is fine, HOWEVER, if I pass either input through the makeup-gain side of the blend board then the signal is heavily distorted. It sounds like the opamps are saturated and signal is too hot.   So when i say it sounds great, I mean when I pass the compressed output of the talkback through the 'clean' side of the blend. Passing it through the other side sounds like crap!

So, since the only difference between each side of the blend board lies with the makeup gain pot setup (R38 or R39).  Can someone check the schematic for the blend board and see if the input makeup gain pot is supposed to be wired up like that...?   Maybe I'm supposed to leave out the 10k resistor on the feedback arm of the input opamp (R11 or R13). I didn't experiment with the makeup gain section yet except I tried removing the 10k resistor upstream the pot (R16 or R18) and also adding another 10k upstream to make it 20k.  I think there's too much gain happening at the input on that one side of the blend board.  The pot I have to work with is 10K linear.  Anyone see the problem?

I was originally thinking of setting up the makeup gain like this (below), but it was suggested to try the other setup. Dunnnnoo
op-amp5.gif
 

Attachments

  • Talkback Limiter Blend.pdf
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SSLtech said:
This way instead:

Redrawgain.jpg


Keith

So, I've looked at the schematic and even duplicated it in a simulator and I still dont see how this pot setup should work.  Am I missing something? If you check out the schematic above,  if I leave out the 10k resistor at the input (ie. R18 or R16) then I can get unity gain with the pot at minimum or hugggee gain if the pot is cranked up. I don't know to change up this setup to make it work to have unity up to 20dB gain.  I could do it if i put the pot on the feedback arm the way I've seen it done and it would work... SSLtech what do you think? Am I doing something wrong?
 
So I did some tweaking on the blend board for the makeup gain opamp.  

input_gain_talk.bmp


Theoretically, if R18 is about 2.2K and R14 is about 12.2K then IC3A will provide 0-20dB gain. I found this from a simulation using NI Multisim.   I tried these values and hooked up the blend board to the talkback board, but the gain wasn't right, I think it was only about 5.5dB gain at max.   So I changed the resistors to 1.1K and 11.1K which would theoretically give about 25.5dB.  It was a bit off so I tweaked the input resistor R18 to about 2.5K. Seems to give 0 to 20dB.  So R18 at 2.5K and R14 at 11.1K seems to give 20dB gain and unity when the pot is turned down.

I guess my problem was that if you have a theoretical gain at an opamp and you put a resistor upstream of the input then your gain decreases right?  I don't know if there is a proper way of 'matching' an input to an opamp.. or how I would have figured out which resistor values i needed analytically.  Maybe this has to do with the 'ground reference.'


Also, on the talkback board, i wanted the input opamp to give 0-20dB gain too, however it gives about 15.1dB gain.

Theoretically, shouldn't this input give 1/1+1 = 2X gain when pot is at min and (1+10)/1 +1 = 12X gain when pot is at max?
(theory from http://www.colomar.com/Shavano/intro_opamp.html)

So 12X gain should equate to 21.6dB gain*...  Which isn't 20dB, but why am i then getting 15.1dB gain hmm....
(* from http://www.muzique.com/schem/gain.htm)
 
jdiamantis said:
geetar_king said:
wired up a channel last night but didn't have much time to play around with it.  it's passing audio! phew!  still some work to do with the blend board.   man this thing really compresses though when it kicks in!  i put it on a snare track and i could almost here my neighbor's microwave coming through in the background  :p

anyone know what the best way is to set the bias trimpot for the FET?

When I made this post, I now realize that the first sentence may be misleading; I was using a mental shorthand when I wrote, it the instructions may not be clear...

Clarifications are underlined and in italics;
Looking on the schematic, you would remove jump 2 so that when you apply a test audio signal the side chain is not trying to compress.

On most FET type compressor/limiters, you first want to adjust the trimpot so the gain through the circuit is at maximum  i.e. no reduction in gain from the fet.

apply a tone signal to the input (1kHZ will do), and monitor the output with a device that can read dB out. adjust the gain of the signal so it high enough to read on the output of U1B, but not high enough to distort.


Adjust the bias pot so you have maximum gain through the circuit. Turning it one way will reduce it; the other will increase. Wait for the gain to settle. Make sure you can't get any more gain turning the pot.

NOW, slowly turn the bias trim pot until the output of U1B drops by 1dB. This may take a little effort as the bias circuit has a long time constant in this circuit. What this does, is turn the FET channel on so it's always conducting whether you are compressing or not. This reduces distortion.

return the jump 2 to it's place, remove the test signal and apply audio.

Compress and enjoy.


Hopefully, this is the way it turned out. Otherwise, follow these clarified instructions.
 

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