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There are a handful of plugins (effects and processors) that have optional usb controllers, but those can quickly clutter the workspace. Plus, DAW plugins automatically recall the session’s settings, rather than notating an outboard units settings. I like you idea in principle, but in practice it takes a certain dedication to a slower workflow. I’m interested to see how this works for your team.
It's true, there are already few hardware controllers for plugins and they tend to occupy the desk, but with a touchscreen solution you don't have to write anything down, they are just plugins as well.

I think the advantage of such solutions lies in the better communication with the musician during tracking, you're not just staring at the computer monitor armed with a mouse, but are much more open to what's happening in the recording room.

Another advantage I think is the fact that, especially in a band (rehearsal/demo recording)context, they usually work without a dedicated operator/audio engineer, so simple tools for controlling the rather static recording situation are a real advantage.
 
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Secondhand/inexpensive tablet, running some KVM software? You'd need wifi on the host, of course. I can't think of any showstoppers at the moment, and I'll be doing something similar, but I haven't actually done it yet.
That's a good idea, I've already thought about it. An old iPad or similar comes with everything it needs!

I even still have an old iPad 3, but unfortunately it's already in permanent use as a sound generator on my master keyboard. Good old Garageband still sounds convincing after all these years... ;)
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I’ve been working on a cover of Laura Nyro’s the Wind with my partner and I’m always floored by the sound of reverb on 60’s recordings. I’ve got a cheap lexicon I don’t use very much and I also have some Waves Abbey Road plug-ins I like until I’m actually using them.

I find it very difficult to achieve the level of lushness and presence yet somehow not drowning in it like in recordings of Laura Nyro, the Beach Boys, or Brazilian stuff like early Gal Costa.

I’m sure they had access to amazing plates and probably rooms designed for certain sounds, but for some like me who has a studio full of DIY equipment and a computer, how can I get close to that sound?

Is there any reverb builds that y’all know of? A plugin many recommend? Or should I just set up a speaker and a mic in my garage or something? ;D
It's not just the chamber, but the level of the signal hitting the speaker in the chamber, and the gain of the pre recapturing the verb. If you listen to those old Beach Boys the reverb is super saturated. Part of what makes it lush is saturation. I've had good results just using a big room (or small, doesn't matter that much). Speaker on one side, mic on the other, and CRANK the speaker aux. Then on the capture mic, crank that gain. Use a compressor on your capture, shelf the nastiness out of the high end. Another thing I find useful is then compressing the vocal or instrument in series with the chamber capture, so they don't sound like separate entities (Rather than using something like an aux send for the reverb).

For the Brazilian style verb, pick up an old Ross mixer from the 70's or with a built in sprint reverb. People give them away.
 
Nice thread! Good suggestion on vst’s I didn’t know about.


(A bold statement but if we suppose: the perfect / most natural system of artificial reverb doesn’t excist… the following would be fun to mess around with):

To the best of my knowledge is that one thing to define a realistic reverb is that it’s mostly in the early reflections. There are some papers on the net reflecting this with measurements. It’s very hard to calculate the ER of rooms. This is only logical because that’s the part of the reverb on the hitch we can define a localization. The (long) tail is much more diffuse.
-> modulating (examples: panning, slight frequency shift, HPF, dynamically different, adding tonal resonances) the send signal that goes to an reverb with only ER part of a plugin that goes to another bus with only the tail / diffusion part of a reverb plugin (which also gets the channel signal).. this way we could mix different verb. plug-ins to make a more realistic or carefully designed reverb sounds.


Also both are dynamic and not static (as in an IR.) good convolution and algoverbs deal with this principles/information on the bases of the sends panning. Which is most likely a made up setting with variables that conclude in a non linear frequency response.
-> keep in mind the position of the players vs. the listeners position vs. Placement in the room. There are some free IR calculators that let you design a room with different players / (speaker) positions. Since it’s not close to a real ER we could approach this as an in between reverb part for a more realistic panning. (First ER, as above mentioned - Panning ER IR+ Tail)

Then there is the phenomenon of room-modes . The thing that could happen in various sizes is a pitch depending frequency based reverb time. Thus if a note is played short the reverb time is shorter then if that note is played longer. There are different types of room-modes and the release shape and summing (+ kinda feedback) plays with some patterns we could only approximate.
-> A fun thing to do is using a “cepstrum lifter” (for the unknown: this is kinda the fft of an fft) with that tool we can set a frequency + length threshold and for example filter the rest out and put the residu signal tru some extra apf, resonators or rt granular stages before sending it to the above kinda reverb busses.

One thing to keep in mind (if you like / may depend on genre) is having a phase coherent sum… or not especially in the ER phase in combination with your source. Sometimes I like to use transient or body depend dynamic eq (or multiband compression like dmg) ducking the clashing or masking frequencies in the reverb part a bit and thus stimulating the air pressure movement which is stronger from the player.
-> on that note.. having a transient splitter cut or reduce the transient thus sending the the body (in an 3 or 4 way linear xo path) will help with realism a lot. Also I like to give the left and right signal different values for the attack time. This again will give (imo) a more natural placement in the simulation room (or at this point imagined room)

Just like almost anything else we do in mixing there just to many variables to come to conclusions or one way solutions.
-> and just like many audio processing things it I find it nice to send the whole sum of the above to a couple of busses. For the last bit of dynamic control/ eq / saturation etc. to glue the things together. Stereo and coherent analyzer could help/navigate here for mono and angle compatibility. Some modern stft plug-ins could help here. For example a “reverb remover” algorithm can give us more parameters to play stylistically if we like. For example there are stft coherent extractors and splitters, ssb modulators and more. (Fun to play with!! But I could also highly recommend some older fundamental Blumlein papers and stereo principles for getting realistic stereo image (freq vs td and ld curves))

maybe long post.. but those are somethings I found useful in dsp. It gives me a lot of times such more dynamic depth in the soundstage image. And ofcourse it doesn’t have to stay itb but problems we face then are also nice and endless but I could share some thoughts about that + the combination of the above another time if people would like that.
 

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