Console signal flow, input cards, general discussion on API styled DIY mixer

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One more thing boji.

There is a rounder sound going through the channel amps and transformers.  That can get murky on some inputs.  A lot of the times today I use the Preamp out on the Patchbay. Direct to the recorder input, (protools) for a cleaner or natural sound.  Usually I use preamp out on vocals and Acoustic instruments excluding drums which I like the fatness of the low end from the Transformers or what ever the extra circuitry does. 

My point is If you have a patch point for the mic pre out and thats all you need, you may not need the relay/switch.  Just patch the preamp direct. 

Ive also thought about building an input to the main API mix bus for Protools out. so as to use the outputs from the 192 I/0 straight into a passive interface to the main mix bus, avoiding all channel amps and having the input channels open for using on other things like my Analog Recorder  or mic pres to send to tape for tracking.  This becomes a simple passive summer with the least amount of signal processing for a Dangerous Summing  type of circuit. That's for when you want passive summing and in the box mixing.
 
rounder sound going through the channel amps and transformers.  That can get murky on some inputs

Why do you think that is? Are you in need of a recap?

you may not need the relay/switch.  Just patch the preamp direct.

Id rather not use break on insertion jacks for my main signal. Patch panels will certainly take care of effect insertion and the occasional odd rerouting, but I'd rather not plug and play every time I'm switching between recording and playback. I'd also like to use the input cards to add some extra iron in the path when recording.
 
Experienced recording guys, I'm thinking of using a single solo button to do a few things based on the way you interact with it.

For example:

quick press- standard solo
long press - grouping
two quick taps - PFL

Do you see any problems with this arrangement? 
The only issue I can think of is if you want to pop on and off a solo channel in a musical sort of way, but you could not because it would initiate the PFL. An arrangement in the DAW would be the workaround of course. But is there any other monitoring situation that would conflict with this button logic? 
 
The reason for the rounder sound is you have more transformers in the signal path.  If you think about it each transformer is a bandpass filter and it imprints its sound multiple times depending on the path choosen. 

Preamp out straight to input of tape/daw is the least amount of tx and amps/bandpass filters, to the recorder.  There is an extra transformer in 550 eq, and a trans in the channel fader, and another in the sum amp for bus ACA, and another for the insert to Pgm bus 1 thru 8 sum out.  The more transformers you go thru the more change in tone.  That 6 to get to tape or as few as 2 depending on the path.

I have replaced caps on some channels and found no difference in sound for those channels.  So if I have DC offset or bass roll-off or distortion through a Cap I replace that.  But I have only replaced the channels with problems.  I also use a spectrum analyzer to check the signal path for differences between channels. 

Transformer problems goes back to the Tape Days.  If you ran through the whole board twice, Record then playback, you go through a lot of bandpass filters.  That could go from Fat to Muddy sounding.  So when you want transient response,  You patch direct from the preamp out. 

To do routing for inline monitoring, all you need to do is have the line in line out tape/Daw be post EQ and Pre feed the Cue, Echo, mix bus, Channel fader to mix sum.    With the standard routing using channels, you give up having a Fader and hence on an API you had a side car monitor.    Today I'll bet a lot of people use the DAW for the side car mixer and make the channels / preamps go to the input of the recorder.

This potential loose or hard sound is more hard sounding on digital and more loose on tape.  Tape is dead as far as most people are concerned. (Excluding some people on this forum).  So digital may benefit from having a loose sound from going through a bunch of bandpass filters.  You have to listen to decide.    I do think that digital sounds better when you EQ any issues going to the recorder.  When you analog EQ on the way back it can get a ringing quality to the top-end if your not careful.   




 
Using a push button to multi functions is a digital implement.  It may become second nature but it also becomes a thinking function.  I'm not sure I would like that but it might become muscle memory.    The Solo I assume you mean in place solo or do you mean a Solo buss.    In place would mute other channels and disrupt the mix.  Pfl would be to a Solo Bus and not disrupt the mix.  Right?

What is Group?  Is that for monitor input? Not sure what this does.  It could be a couple of things. Or is it a group solo?
 
Group would be group SIP.
Any channels with the same group assignment would follow any action on any solo button that was also assigned to the group.
For example assign multiple channels to group solo. Singularly turn on or off any solo on any of these channels and all group assigned channels copy that channels state.

I have a four channel prototype of the button logic working right now, minus the PFL.
Adding the PFL would supersede the solo and group solo logic with a SIP styled, pre-fader listen. The standard solo (non PFL) would of course be post fade.

I have concerns about tacking on PFL because I am inexperienced with working on large desks- I may have overlooked a practical reason for having two buttons, one for SIP, one for PFL.
 
I did live broadcast mixing on large format.  The pfl was what you used because you were on air.  There was also a SIP Safe override.  You did a mix while laying back to 48 (if a remix is needed).  Usually there would be post production as well.

Solo is important to not damage the broadcast.  On your application it seems workable.  And a lot of bang for the buck,  the other side of DIY.
 
boji said:
I have concerns about tacking on PFL because I am inexperienced with working on large desks-
As fazer says, when you need the mix 100% safe (as in broadcast and theater, but also in music recording, when you feed the musicians with the C/R mix), you want PFL.
  I may have overlooked a practical reason for having two buttons, one for SIP, one for PFL.
I'm used to push-buttons with different behaviour depending on duration of press. I've used them with RTS intercoms, who were the first to introduce them to the audio community, IIRC. And I still use them daily on my Tascam DM4800 desk, and still fail to operate them properly when the heat is on. * They work fine in standard conditions, but in tight situations, I know that  I want the simplest behaviour: momentary or latched there is a muscle memory for that, but momentary/latched/whatever takes brains, which I don't have in these situations.
The idea of having two separate buttons is mine; I don't know if anyone had the same idea and put it in reality, but I know that I would do it if I made a mixer for myself and didn't have to justify it to a paying customer. I would have two different types of push-buttons, located separately enough.

* The talkback buttons on the DM work exactly like that: short push is latch, so you have to push once more to turn it off and unDim the C/R, long push is momentary. Everytime I'm doing something that requires some attention, like trimming a range or setting a punch-in/out point, the anxious musicians wants to hear the sound of my voice, and then, without fail, I'll do the wrong push (if I manage to hit the right button :-[ )

P.S: meantime I watched your vid, nice work, beautiful frame. You being probably much younger (less old ?) than me, assuming you have video games dexterity, you may have more facility for the subtleties of press duration.
 
Hi Boji, would you mind posting dimensions of the metalwork you have? I was thinking of building some racks in a similar style that I could eventually turn into a console.
 
Can anyone help me to the schematics of the 1048 input-module and 510 send-module?
I looked around but did not see these posted on this forum. I'd be interested to see them as well.

My line input will drive the fader if the eq is hard bypassed. This will make fader calibration difficult.  I don't think I'll care very much where 0db is though. I'll just mix with my ears.

Hi Boji, would you mind posting dimensions of the metalwork you have?

No CAD on the frame. To be perfectly honest, I simply drew angles on a fiberboard until it looked pleasing, then made an aluminum template and used a laminate router to churn out the sides.

I can tell you that I made the bays 14" wide so as to hold 8 x 1.5" channels plus 1 inch trim pieces to both sides. This extra length makes room for the VU meters and gives a little room for the input cards when plugging in, as they don't fit without turning them slightly to the left or the right. The input cards are 11" long, another odd size, but that's how the buttons and knobs stacked up.

The size of the 500 slots is darn close to the standard API dimensions found on the webs or within this forum.

I'll measure stuff for you once I get it put up on my table and I begin the wiring of it, but I have to finish the input card prototype first!
 
I briefly looked over the flyer you attached Balijon. All those primary and secondary mutes, 'big" and "small" fader route assignments to me would make a mix process fairly complicated. 

I'm learning to dislike desk layouts that you can not revisit a year later and dial up the sound without consulting a knob/button position chart.
Increasingly I rely on my DAW for complicated patches, sub-subgrouping, ect.

Which is to say I think the more capable our computers get, the easier our hardware ought to be to figure out. I guess this is what Apple has been about.

I do like the active variable hi/lo pass that shares an encoder wheel. That sounds like a fun arduino project down the road; use relays to trip different cap values.
 
boji said:
I briefly looked over the flyer you attached Balijon. All those primary and secondary mutes, 'big" and "small" fader route assignments to me would make a mix process fairly complicated. 

I'm learning to dislike desk layouts that you can not revisit a year later and dial up the sound without consulting a knob/button position chart.
Increasingly I rely on my DAW for complicated patches, sub-subgrouping, ect.

Which is to say I think the more capable our computers get, the easier our hardware ought to be to figure out. I guess this is what Apple has been about.

I do like active the variable hi/lo pass that shares an encoder wheel. That sounds like a fun arduino project down the road; use relays to trip different cap values.

Hi boji,

I was not planning to completely rig this with individual switches, but tend more towards remote controlled and related configuration recall.

Something like this:
http://www.groupdiy.com/index.php?topic=45781.msg580546#msg580546
Maybe the schematic looks a little complicated, as a user-interface it should be easy to 'drag and drop' a module into a signal chain.

Think midi/IPad with a simple user-interface... Maybe a small channel display to show the configuration chains or (my preference)  a TFT with the routing/setup configuration and VU/PPM's per x amount of channels.
Maybe something like the Arduino can be a building block to provide the steering logic for this.

Theo
 

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Just got done with a major audio project so...back to the mixer!

My first PCB boards arrived and (/drum roll) they work!  yay.


http://www.youtube.com/watch?v=eNJGjBTCvAM

dualrelaypcb.png
 
I should mention that these boards will be used for Mute / Solo signals and for circumventing Mic pre Inserts (post bucket 1) and EQ (Bucket 2).

Thank you abbey road d enfer for giving me a hard time about my previous half-assed fix for this. Now the desk will have 'true' signal bypass.

truebypass.png

 
Concerning the use of a relay with spare connections, is there any advantage to leaving them unconnected, or can I double up the switch, and run the switched signal down both channels of the relay?

Would I ever see noise or other odd problems with running the signal across both sides? Thanks for your replies.
 
boji said:
Concerning the use of a relay with spare connections, is there any advantage to leaving them unconnected, or can I double up the switch, and run the switched signal down both channels of the relay?

Would I ever see noise or other odd problems with running the signal across both sides? Thanks for your replies.
Doubling contacts reduces the risk of intermittence due to contaminated contacts, but these relays are now extremely reliable and don't need these precautions. There is no disadvantage either, except that the increased capacitance may affect the cut/x-talk performance, nothing that an appropriate lay-out couldn't prevent.
 
Hi Boji,

I looked at your video with the latching / grouping of switches with the arduino. I have a similar arduino assisted logic and non-latching switches in mind for my linemixer. do you have a layout for the whole desk? do you run everything from one mega ? enough inputs and outputs? I use fet switches, but theoretically I could have a sub board similar to yours, maybe pin compatible...
do you have code available? would like to see your procedures - might even like to chat about the similarities differences in your mixer compared to my approach... any interest in sharing that much?
my buildlog is (updated only sometimes) here: audiomixer.wordpress.com and http://www.groupdiy.com/index.php?topic=48120.0

cheers,
Michael
 
Hello audiomixer,

Yes I've been following your progress. I dig your photoblog and cardboard proto. I'll PM ya...

 

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