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Crafting circuit models using plugins
Old 17th November 2017
  #1
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StoneyBCN's Avatar
 
🎧 5 years
Crafting circuit models using plugins

Hi folks.

This is an interesting topic that some of us have been having fun experimenting with. 2017 has been a watershed year for digital audio, with digital simulations of classic analogue gear and all it's intrinsic properties coming on noticeably.

Gone are the times of a simple band pass-y waveshapers of early digital, in place we now have sophisticated algorithmic models of analogue-measured current and voltage saturation, RC time constants, summing behaviours, dynamic filters... It seemingly never ends. So, in our quests to live up to the standards of our most respected and admired peers, where do we begin with applying these available processes to our audio in a digital age, where user intent far outweighs the necessities that analogue imposed?

Some of us have been discussing how some of the tools we now have available to us, can be used at will to apply some specific, desirable asset of analogue behaviour to our favourite digital processes. For example, combining impulse-based convolutions with specifically-chosen modulators and saturators, to emulate a particular (or user-imagined) analogue characteristic.

Does anybody know any other cool plugins or tools that can be used to alter time constants, step response, slew rate, transfer functions, or asymmetrical clipping? I've found plugins that do so many of these things, but other things don't seem to be implemented in digital much. So, when I've had to, I've used tools to "hack" stuff to access these processes. Like using a constant DC bias signal to make a stock DAW saturator produce second-order harmonics... A favourite of mine on weak bass signals especially. Once you start building your own headroom and time constants into it, you could easily have designed and modelled a tube or transistor stage, or some crazy new component that has never existed before, but does now simply because you needed a certain job done and only had certain tools at your disposal.

So, what about you? Anybody else tried stringing together complex "analogue" circuit behaviours out of utilities and plugins? Any success, or curiously noteworthy findings?

Half of this terminology seems lost in the digital world, yet much of it still proves very useful in the efforts to introduce certain fidelities to our work. It may also help us to scrutinize the products we seem to invest so heavily in, which as consumers we are entitled to do. Honestly, putting some of the big name plugins under an oscilloscope and seeing nothing interesting happening to the waveform at all, was quite an eye-opener for me.

Therefore, opening some discussion and collaboration may help some us move forward and bridge that gap, the one between I would appreciate if the tone of this thread could stay informative, interesting, and helpful at all times. This is not about bashing developers or talking down to the less informed, it's about the pursuit of quality audio.

Keen to hear some thoughts!

Peace.
Old 17th November 2017
  #2
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🎧 5 years
One example I have had great fun with just before starting this thread, involves combining some awesome Airwindows plugins with Alex B's N73 for Nebula, an EQ library based on the beloved Neve.

The original circuit isn't as simple as my example, but it definitely made the right suggestions:

NC-17 > TransDesk > N73 (no saturation) > Desk4

The idea was to get the transformer/transistor based input amp, feeding the EQ, which feeds a final amp/transformer stage. The EQ is set to clean, because it's interacting with the saturating output stage. One developer's excellent, unique saturation tools and another's super-smooth EQ filters.

I've also really been enjoying combinations of Airwindows Powersag with CDSoundmaster Virtual Tubes, and CDSoundmaster R2R tape machines with Airwindows ToTape... lots of fun, and a more straightforward emulation.

Oh, one more... Altiverb > Airwindows Loud. Made for each other.

Everybody loves a bit of FabFilter Saturn, SSL X-Saturator, Softube Saturation Knob, etc. How are you all using these in your mixes?
Old 17th November 2017 | Show parent
  #3
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I won't get too technical here as I'm not totally breaking things down into specific circuit components, but this is a really good way to think about things. I think I'll do this more. Really interesting topic.

I'm using very clean and simple preamps, so I usually like to start with something that models a Class-A gain stage with transformer .... It's usually a mic pre in Nebula. I'm finding the brand new BRA pre73 plugin to be nice in giving some Neve-style weight, but the great thing is that it can be pushed to a sweet spot of subtle saturation to round off transients and introduce some harmonics back into the signal .... From there, it's a tape emulation to simulate tracking to tape, including the frequency response and further harmonic response. I often use Tim Petherick Nebula tapes for this. And then I'll usually use a Line In emulation to simulate running each track back into a console.

What I'd really like to figure out is the summing behaviour to a pentode tube bus path. How can I best model the way a tube stage will handle intermodulation distortion from all the frequencies in all the channels? Is it enough to slap on a tube sample for the colouration and harmonic generation? Right now I'm using Prime Studio's tube summing plugin Charly for this purpose, and often Gemini Audio's KultComp if I want additional harmonics simulated from the output stage.

Last edited by rosewood123; 17th November 2017 at 07:26 AM..
Old 17th November 2017
  #4
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🎧 10 years
There're so much more to model besides abstract "class A" and "a transformer"... what kind of semiconductors? What biasing scheme? Cascode? SRPP? Feedback network? Not to mention of all small details, like parasytic capacitance, temperature stability, etc...

Even SPICE is not so close to physical world sometimes.
Old 17th November 2017
  #5
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🎧 10 years
Hi pal!

This is the most interesting topic in years, so glad you share this gold info, you have tons of knowledge and I think everybody is going to learn!
Old 17th November 2017
  #6
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🎧 10 years
I'm trying to figure out the worth of BitShiftGain in my signal chain these days.

I use very different modules during my tracking/editing/mixing/premaster stage.

One thing that helped me lot aside from my AA and N4 3rd party libraries is Goodhertz Wow.

It really help revive some of those human feel and none linearities in the mixing stage after automation.

My otario is straight out of maintenance and there's not much going on on this side of the tape effect now. Its a tank.

I'm mixing Hybrid and mostly OTB but im trying to figure out a way to make a simple pass in my consol without resumming 3 times so after my tracking and editing I'm trying to just have a one pass to my board and back into 4 stereo busses in my daw.

The challenge now is that its still too heavy for my poor pc but the result is well worth it. Usually deeper then my OTB rendering but never as warm and wide.

I see the potential but theres really something missing somewhere.

The weight is really absent if I stay ITB and If I fake the funk it just sound muddy loose all this depth ive created which is not the case OTB but with a way higher noise floor.

So i'm wondering If i should invest in my console to get cleaner pre or keep on diging and loosing time ITB to throw all my gear through the window

Last edited by Martel80; 17th November 2017 at 04:33 PM..
Old 17th November 2017 | Show parent
  #7
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Quote:
Originally Posted by zvukofor ➡️
There're so much more to model besides abstract "class A" and "a transformer"... what kind of semiconductors? What biasing scheme? Cascode? SRPP? Feedback network? Not to mention of all small details, like parasytic capacitance, temperature stability, etc...

Even SPICE is not so close to physical world sometimes.
Yeah, totally, which is why I sort of prefaced with, "I'm not breaking things down into specific components." It can get way, way more specific than what I said, and I'm really no circuit designer, but there is always a starting point and the decision to learn. :-) I think it would be really interesting to be able to break things down to lower level design like you're talking about. Sure it's still modeling, but that level of control and design could be really fruitful and fun ....
Old 18th November 2017
  #8
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🎧 5 years
Subbed!
Old 18th November 2017 | Show parent
  #9
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StoneyBCN's Avatar
 
🎧 5 years
Quote:
Originally Posted by zvukofor ➡️
There're so much more to model besides abstract "class A" and "a transformer"... what kind of semiconductors? What biasing scheme? Cascode? SRPP? Feedback network? Not to mention of all small details, like parasytic capacitance, temperature stability, etc...

Even SPICE is not so close to physical world sometimes.
Thanks and welcome

This is really kind of my point. I believe that every time "mojo", "vibe", and "3D" are used as quantifiers, a small piece of the future of audio engineering dies. I'm not an electrical engineer by any stretch, but I've made it a point over the past few months to acknowledge the importance of understanding AC circuits in relation to waveform behaviour. I've got a long way to go and will take all the help and advice I can get, since it's not exactly a simple topic that one can master from late nights on Google.

The point here is to encourage each other to improve our grasps of these concepts. We've all met musicians that can play wonderfully, but have barely any grasp of musical language and theory. Power to them! However, what really worked for me personally was learning music theory and applying it to my playing and composing. The little that I've learned about electrical circuits and components so far has similarly made a huge difference to the recordings and mixes I work on.

It strikes me that I must surely not be alone in seeking this knowledge, so here's a place we can come together and share ideas, help fill in some blanks for each other, and in particular, implementing some of these things ITB.

If you happen to have any interest in these discussions, your level of knowledge would be more than welcome
Old 18th November 2017 | Show parent
  #10
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🎧 5 years
Quote:
Originally Posted by rosewood123 ➡️
I won't get too technical here as I'm not totally breaking things down into specific circuit components, but this is a really good way to think about things. I think I'll do this more. Really interesting topic.

I'm using very clean and simple preamps, so I usually like to start with something that models a Class-A gain stage with transformer .... It's usually a mic pre in Nebula. I'm finding the brand new BRA pre73 plugin to be nice in giving some Neve-style weight, but the great thing is that it can be pushed to a sweet spot of subtle saturation to round off transients and introduce some harmonics back into the signal .... From there, it's a tape emulation to simulate tracking to tape, including the frequency response and further harmonic response. I often use Tim Petherick Nebula tapes for this. And then I'll usually use a Line In emulation to simulate running each track back into a console.

What I'd really like to figure out is the summing behaviour to a pentode tube bus path. How can I best model the way a tube stage will handle intermodulation distortion from all the frequencies in all the channels? Is it enough to slap on a tube sample for the colouration and harmonic generation? Right now I'm using Prime Studio's tube summing plugin Charly for this purpose, and often Gemini Audio's KultComp if I want additional harmonics simulated from the output stage.
Hi Rosewood! Glad you jumped on board.

Sounds like we are definitely thinking similarly here. I'm in a situation at home where I'm down to using a Focusrite interface for a lot of things, and once in a while I put my pennies in a pot and go to my local studio to use a few of their things. It's a writing/arranging pre-production thing that I've been working on a while. I'm using lots of SSD4, Kontakt libraries, amp/cab sims, Altiverb, all the things I can get my hands on to "fake it" until the project is ready to invest studio time into. I decided that I'd like to get the best out of not just the songs, but the gear too. So, as you say, I decided to learn

Regarding the Class A element. If you're using just a stock DAW distortion utility, and assuming the coding is functioning correctly, at first you might assume this is doing a typical Class A saturation (where the + and - sides of the wave have an equal amount of headroom). However, if you get a huge peak on the positive side and less peak on the negative side, and they both get clipped equally despite their individual amplitude, then that is not really reflecting most Class A component behaviour. Well, saying this, I could be totally wrong - just sharing ideas here - but I think true Class A in analogue would be clipping either side of the wave INDEPENDENTLY of each other.

This is a behaviour that I need to study more to truly be sure, but nonetheless I've found a way to do it that requires a stereo input saturator, a duplicate of your signal, and two opposing DC voltages feeding the signals into the stereo clipper, then sum the output to mono...

Thing is, there a certain guy who provides tools that can do this ITB, incredibly well IMO, but the DC part is quite dangerous to your monitoring setup and requires preventative measures (like a DC filter on the 2bus, for a start...)

A pentode tube? Hmmm. That would be a task indeed, at least for me. I know some basics about triodes, no idea how a pentode may differ in audio applications. However, I'm sure with some very creative routing and lateral thinking, there are probably some tools that could be used in that way. One to dwell on...
Old 18th November 2017 | Show parent
  #11
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🎧 5 years
Quote:
Originally Posted by babiuk ➡️
Hi pal!

This is the most interesting topic in years, so glad you share this gold info, you have tons of knowledge and I think everybody is going to learn!
Hey amigo! Great to see you here.

Some of our discussions got me thinking a lot on the subject, and we now find that there are others just like us that want to understand all this analogue stuff really means. You also had some great ideas about creating a kind of live recording session mentality even when using libraries and emulations. I think that is another excellent subject, maybe we can bring some of those ideas into this
Old 18th November 2017 | Show parent
  #12
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🎧 5 years
Quote:
Originally Posted by Martel80 ➡️
I'm trying to figure out the worth of BitShiftGain in my signal chain these days.

I use very different modules during my tracking/editing/mixing/premaster stage.

One thing that helped me lot aside from my AA and N4 3rd party libraries is Goodhertz Wow.

It really help revive some of those human feel and none linearities in the mixing stage after automation.

My otario is straight out of maintenance and there's not much going on on this side of the tape effect now. Its a tank.

I'm mixing Hybrid and mostly OTB but im trying to figure out a way to make a simple pass in my consol without resumming 3 times so after my tracking and editing I'm trying to just have a one pass to my board and back into 4 stereo busses in my daw.

The challenge now is that its still too heavy for my poor pc but the result is well worth it. Usually deeper then my OTB rendering but never as warm and wide.

I see the potential but theres really something missing somewhere.

The weight is really absent if I stay ITB and If I fake the funk it just sound muddy loose all this depth ive created which is not the case OTB but with a way higher noise floor.

So i'm wondering If i should invest in my console to get cleaner pre or keep on diging and loosing time ITB to throw all my gear through the window
This is an awesome post! There are a few things here im going to stew on for a bit because there is a lot I'd like to ask about your setup. Really appreciate the contribution here.

For the time being, I'll ask if you've tried Chris's bit/sample shift test? I haven't, but it's very interesting for sure.
Old 18th November 2017
  #13
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🎧 5 years
Totally don't mean to hog the thread, just checking in before I get busy with something else. I'll post a tutorial here for how I use test tones and an oscilloscope to test plugins and supposedly design my own "circuits"

Stay tuned.
Old 19th November 2017 | Show parent
  #14
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5 Reviews written
🎧 10 years
Quote:
Originally Posted by StoneyBCN ➡️
This is an awesome post! There are a few things here im going to stew on for a bit because there is a lot I'd like to ask about your setup. Really appreciate the contribution here.

For the time being, I'll ask if you've tried Chris's bit/sample shift test? I haven't, but it's very interesting for sure.
No I havent.

I gain stage everything because theres no need to keep a mix balance unless im actually mixing as long as I keep the focus on where im going.

Do you have any experience you can share with this AirWindows BitShift gain ?

What is the difference with my standard gain plugin ?

Is it noticeable ?

And to what extent ?
Old 19th November 2017
  #15
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🎧 15 years
When we get into these analog modeling concepts, I think one of the problems would be to establish a connection between a distinct attribute of sound, and a theoretical concept or process that takes place within analog hardware circuit.

Practically, everything networks with everything, but I believe it is useful to distill some of the important principles. Like, it is not difficult to learn how "clipping" sounds with different types of music and instruments.
But this is endless, and we need to find at least a certain number of these concepts, to recognize, recall, and apply.
What I mean here, it is also a very individual thing, because the hearing abilities and understanding of "sound" is different among individuals. This becomes obvious, when we talk about details in sound production, e.g. among the band members.

To me, during learning and studies, it is important to work with "nuclear" devices in the sense of there is one core feature, and not a myriad of "interdisciplinary" knobs from drive to frequency to duration to "phatness".
Then, we can combine these devices to create a certain color and attribute in the sound, that we design for aesthetical reasons.
(While the pure technical issues like unwanted bump in the spectrum or too wide dynamics are much simpler to learn and apply.)

I'll share one particular process that I designed as a freeware replacement for various commercial and delicate bass enhancers. The purpose is to create a very versatile and efficient bass harmonics synthesizer.

The chain so far, there are variations per song..
1) optional: NWEQ or another additive EQ, to enhance frequencies in the original source that should be processed, but somehow are starving. You may use some higher Q factors here, but avoid ringing.
2) a very soft sounding but deep compressor. I'm using the free, grey Klanghelm version with timing set to AUTO. One purpose would be to keep the saturator within a sweet spot, and to avoid crackling or thumping sounds.
3) MEqualizer as an input filter, it uses the steep hipass and lowpass filters (e.g. 48dB/oct) to restrict the frequency band to what should be processed. Above 400Hz, I need nothing. YMMV. Depending on the played notes, and we could use automation, the band stop might be as low as 200Hz or a bit lower.
4) Melda Saturator, it changed its name, was also called Limiter, but in fact it is a saturation and harmonics synthesizer. I created a complex preset for this.
Parameters for harmonics naturally must be set very high. Second and third harmonic is interesting, for hardness and rock feel you might use the higher harmonics knobs also. Use a very round saturation curve, and drive it somewhat above the center, or higher for harder sound.
5) Output filtering with MEqualizer, with a steep highpass we remove the 0th harmonic as much as possible. So this might have a slope from 80Hz to 150Hz or something. A major problem here is phase issues. Then, with a lowpass, also very steep, we remove everything beyond 250Hz, or for some sounds we set it even to 500Hz. 24-48 dB /oct.
Settings of the output filter are very sensitive to the color and balance of the resulting sound. We can use a couple of bell filters to adjust critical frequencies up or down.
6) Perhaps a phase rotator, like what Airwindows is offering.

This is all stacked in a parallel channel. You add that effect to the original signal, and have to take into account there will be phase issues. But it can be handled, and the bass sound becomes extremely fat, so it works also well with the notorious smartphone earplugs. Please, no overdose.


So, this would be an example where we have all important aspects of the desired sound synthesis materialized in certain distinct devices.
I guess it's called engineering.
Old 20th November 2017
  #16
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🎧 5 years
Cool posts, im enjoying this.

Ok, for those that might be interested. Here's a good way to measure and analyse plugins that supposedly model analogue behaviour.

I use a tone generator (one in most any DAW) and start with sine waves. Split the sine wave out to two channels, one panned hard left and one panned hard right. Put your effect plugin on only one of these panned channels.

Use J Scope by Jagged Planet in your master bus. It's a freeware VST easily found on Google.

It analyses waveforms coming out of L and R, so in this case we compare the clean sine to the distorted one. It has a really cool I-V curve setting too, so you can analyse transfer function differences. Have a look at some lissajous curves for different analogue components (I'll post some pictures later).

Here are some plugins to try:

Slate VCC (various analogue circuits)
Any Distressor emulation's THD stage (modelled FET stage)
Slate FG 401 and 1176's have various transformers - turn the compression off first...
Any Neve or API emulation worth its salt should do a bunch of transformer curves
Neve' s also will have transistor properties - slightly asymetrical clipping
API's will also have Op amp properties, the square wave step response should change with frequency
Some Softube, Soundtoys and Kush behave wonderfully under the scope. Some of the "prize plugins" of the scene don't do much at all.
Things like Airwindows provide individual "pigments" which you can combine to make your own desired circuit response. You can get real creative with routing (feedback networks, bias currents, crosstalk...) - it's fun, if this is interesting to you.

Maybe some comparisons could highly benefit from this method. Maybe we should shoot out some various component models on the market, for enlightenment purposes

I do it a lot in my down time and believe me, the results can be very interesting indeed.

Really cool for sorting the men from the boys in the crowded, noisy world of analogue emulation...
Old 20th November 2017 | Show parent
  #17
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🎧 10 years
Quote:
Originally Posted by StoneyBCN ➡️
This is a behaviour that I need to study more to truly be sure, but nonetheless I've found a way to do it that requires a stereo input saturator, a duplicate of your signal, and two opposing DC voltages feeding the signals into the stereo clipper, then sum the output to mono...

Thing is, there a certain guy who provides tools that can do this ITB, incredibly well IMO, but the DC part is quite dangerous to your monitoring setup and requires preventative measures (like a DC filter on the 2bus, for a start...)
Chris from Airwindows? I was taking a look at that plugin a couple days ago and just decided to avoid it until I know what I'm doing with it.
Old 20th November 2017 | Show parent
  #18
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🎧 5 years
Quote:
Originally Posted by rosewood123 ➡️
Chris from Airwindows? I was taking a look at that plugin a couple days ago and just decided to avoid it until I know what I'm doing with it.
Smart move! It's basically gonna give you that super loud 'pop' you get from accidently unplugging something... which is liable to melt your monitors if precautions aren't taken. The reason I won't go much further into it, is because I can't be liable for that happening!

I will describe a few tricks I've found at some point. Also there are some excellent freeware guys worth a mention here. But I do like to direct people towards a certain Patreon, for the future of all audio-kind...
Old 21st November 2017 | Show parent
  #19
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🎧 10 years
Quote:
Originally Posted by StoneyBCN ➡️
This is really kind of my point. I believe that every time "mojo", "vibe", and "3D" are used as quantifiers, a small piece of the future of audio engineering dies.
I don't mean to interrupt your thread, please don't even reply, but I had an experience the other day where I really experienced 3d sound and 3d collapse.

I was doing some critical listening, trying to winnow down some of my tools. Put a plugin on a buss, and when I engaged the plug, the sound field completely collapsed. Without the plug, there was width in the buss, the bass was centered and heavy low in the middle, and there was a high shimmer above that, so there was a sort of pyramidal sound.

.....^.....
_/..0..\_

Hope that diagram makes it through htmlification.

But when I engaged this plug the sound stage turned into this:

=====

I've taken it for granted that the golden ears mean what they say about 3d sound, but this was my first experience really hearing this type of total 3d collapse, it was so dramatic.

I'm not an EE, I can't tell you what the plug did to cause the collapse, but I definitely could relate my experience with the plug to someone who has experience with 3d collapse, so 3d sound does have a communicable meaning, and is thus a valid sound descriptive.

In regards to your other verboten words, if you've experienced it, you know it. You can learn how to get it, or how to not lose it through the process. So they remain useful terms for me, and I suppose the other people who use them. Now if it is in marketing wank, yeah, whatever, any ad copy I read just looks like this to me:

"Lies, lies, lies, lies, really big lies."

But when I'm working, I know when I have the mojo going on. Some people's natural instinct is to quantify everything, hammer all round pegs into square holes. I can't stop you from doing this, and I really don't care if you choose to. I'd just ask that you not denigrate people who effectively use these terms to communicate their experiences in sound.



It's all good.
Old 21st November 2017 | Show parent
  #20
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🎧 5 years
Quote:
Originally Posted by nowaysj ➡️
I don't mean to interrupt your thread, please don't even reply, but I had an experience the other day where I really experienced 3d sound and 3d collapse.

I was doing some critical listening, trying to winnow down some of my tools. Put a plugin on a buss, and when I engaged the plug, the sound field completely collapsed. Without the plug, there was width in the buss, the bass was centered and heavy low in the middle, and there was a high shimmer above that, so there was a sort of pyramidal sound.

.....^.....
_/..0..\_

Hope that diagram makes it through htmlification.

But when I engaged this plug the sound stage turned into this:

=====

I've taken it for granted that the golden ears mean what they say about 3d sound, but this was my first experience really hearing this type of total 3d collapse, it was so dramatic.

I'm not an EE, I can't tell you what the plug did to cause the collapse, but I definitely could relate my experience with the plug to someone who has experience with 3d collapse, so 3d sound does have a communicable meaning, and is thus a valid sound descriptive.

In regards to your other verboten words, if you've experienced it, you know it. You can learn how to get it, or how to not lose it through the process. So they remain useful terms for me, and I suppose the other people who use them. Now if it is in marketing wank, yeah, whatever, any ad copy I read just looks like this to me:

"Lies, lies, lies, lies, really big lies."

But when I'm working, I know when I have the mojo going on. Some people's natural instinct is to quantify everything, hammer all round pegs into square holes. I can't stop you from doing this, and I really don't care if you choose to. I'd just ask that you not denigrate people who effectively use these terms to communicate their experiences in sound.



It's all good.
Hey, thanks for bringing up some points here I admit I probably overlooked.

I don´t intend to deride anyone at all for subscribing to a certain attitude or mindset, nor really is there any fault in developers catering directly to that mindset. If good music is getting made, then it´s all honey in the pot

My feeling however is that, due to these market-driven products being so successful in catering to the "sounds good/is good" way of working, there is a big knowledge gap among us in the DAW generation. Of course, the hard truth with audio is that the "sounds good/is good" rule is always the one to live by. If that´s working well for anybody, then why change?

On the other hand, I don´t necessarily agree that the intention in these kinds of experiments and discussions is for satisfying needs to measure and quantify. Not quite so, at least. When I used to teach, and I´d help someone get one of those "Aha!" lightbulb moments, I used that moment to validating the learning process - that when you understand something, you have access to it´s power.

So, I´ll be a bit clearer - my intention with this stuff is to try to encourage more of this knowledge in the digital age, so it doesn´t become as much a relic as the gear itself. I don´t know if people try to keep this stuff like a closely gaurded secret, or if perhaps many of us simply having thought about it much yet. I haven´t seen any promo materials banging on about having "the best emmitter follower behaviour in digital yet" or WHAT that even means, and yet the products are evolving so fast exactly because devs are starting emulate exactly this type of behaviour. I think it´s as valid a debate how good the Distressor FET stage is being emulated as much as it´s various compression curves... and if the devs realize we are on to them, they would be well advised to stay at the top of their game

Anyways, enough from me. Will start posting some findings soon. That´s when this starts to get interesting. Sorry for the response, but thank you for being diplomatic Nice doodle too! Came across well for me!
Old 26th November 2017
  #21
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🎧 5 years
"Here's one I made earlier"...

Okay folks, here's a really cool concept I stumbled upon last night - create your own delay plugins.

I tried this in an attempt to create my own customizable Tape Delay plugin with my favourite tools. The results were so good, and the process so fun, that I thought I'd share here, since this gives us a circuit-level access to a delay-based effect that can be customized at any stage, in any way you want, and requires the build of a circuit of sorts within your DAW's routing network.

I did this by taking advantage of three specific tools:

1. Internal feedback routing, which Reaper has
2. Any channel time-alignment tool
3. Any gain tool

As is often the case, I'll have to add a DISCLAIMER about the dangers of internal feedback routing. If you send audio through this without taking precautions (in this case, the gain control in the feedback loop should ALWAYS be lower than the input signal) you might experience huge volume spikes of screechy noise. Still interested? Great!

There are probably plenty of ways of doing this, so I'll just explain what I did:

Create four channels -

1. Input
2. Loop In
3. Loop Out
4. Output

Now the routing:

Input: send to "Loop In" and "Output"
Loop In: Send to "Loop Out"
Loop Out: Send to "Loop In" and "Output"
Output: This is the only track sending to your monitors

What we have is, the input (dry) signal is sending to the the output, and also sending to our little feedback loop. The loop is also going to the output. This gives us a combined wet/dry signal at the output stage, which you will see later we can do cool things with. *Important that only the Output channel is being monitored.* The rest are just sending to eachother internally and never go to the master bus. Just follow the routing above.

Now, the fun stuff:

On "Loop Out", we delay the signal using a channel latency tool. Not a delay plugin, just a plugin that can move the channel back or forwards in time. Our delay loop is already set up to repeat endlessly, so we just need to control the difference in time between our "wet" and "dry" signals. I use Reaper's "JS Time Adjustment Delay" tool to offset in measures of miliseconds. 1000ms is a good starting point.

Finally, on the same "Loop Out" Channel, put one gain control. Any stock DAW gain control is fine. The trick is to lower this gain, so the repeated sounds get reduced more and more as they keep going through the loop. Keep lowering the gain on this control, and you will hear the repeats fall off faster. Essentially, this gain is now your "feedback" control as per any other delay plugin.

Send something to the input, dial in your levels on PurestGain, and your time on the Time Delay tool. Voila.



Most DAW's should also allow you to collapse those channels into one master track, of which I think the Output channel makes most sense. It's handy though to keep access to the sends on Input, for controlling the wet/dry amounts appearing at the output.



To take this concept further, here's what I decided to do. For the feedback gain task, I used an instance of Airwindows PurestGain. It's just a simple volume tool for those that aren't familiar with it, and it's free VST/AU thanks to the developer's awesome Patreon project which I strongly suggest supporting, if this works for you. It has two controls - the Gain control will adjust the volume of your repeats. Set it where you like it. But the "Slow Fade" control, when reduced from it's default position, will do a smooth fading of volume that for whatever reason sounds nice here. I was trying to make a convincing Tape Delay type tool for an Aux. I used an awesome Airwindows tape emlation called Iron Oxide 5 on "Loop In", but I suppose you could use anything from anyone. I like IronOxide5 for it's bandpass tools and grungey type of tape. I also added TapeDither after it for some additional tape noise. I also put TubeDesk on the "Input" and "Output" channels, to emulate some analogue amplification circuitry in logical places. Again, choose your own poison. I messed with the stereo routing in the feedback loop to create some stereo crosstalk... the repitions get more and more mono, just a touch Finally I added Console4 Channel to the "Input" and "Loop Out" and Console4Bus on the "Output" channel for some hi-res summing. Very nice finishing touch.

Of course I could have just made a feedback loop and stuck Slate VTM or anything else on it, and be done. But I wanted the repeated passes through the tape to be seperated from the amplification section (in this case, the "loop" is literally acting like a loop of tape), so I used the in/out sections to flavour to taste, at circuit level.

I experimented with a few other things, like a bitcrusher in the loop that will continually melt the audio into odd new shapes... ... Different panning ideas, messing with polarity, stuff like that. Not exaclty circuit modelling but rather delay modelling, but since you're starting with pure clean digital DAW and have access to the full inner workings of the thing, it's all at your own mercy to customize into whatever kind of delay you want it to be, at circuit-level. You're not bound by modelling any particular unit, but rather you can model the unit into whatever you need it to be.

This negative feedback thing is really important for certain circuits. It's interesting to control saturation by using negative feedback the way it is used in real circuits, and I think that merits someexploration. It has also proved to be very useful for me in making my own reverbs and rooms, using similar concepts to the above but with things like allpass filters and crossover behaviour. That's... a whole other bag

Anybody try anything similar before? Or maybe ways of taking these concepts further?? Let us know.
Old 1st December 2017
  #22
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🎧 5 years
Frequency Distortion Modelling - Gain vs Frequency

So, let's get started, shall we?

Distortion.

This word means something is distorted. It's a concept I don't think most people battle with

For audio purposes, we could say this: something goes in the box > something comes out the box, but different. Distorted, actually, from it's original form into something else, thanks to all the lovely gubbins inside the box. Transformers, tubes, capacitors, resistors, transistors, you know the stuff I mean, of course.

If the process happening inside the box gave us a simple 1:1 of the original, according to several defined and measurable parameters, we'd call it a linear process. Obviously, there isn't anything THAT perfect in the real world, especially all those little gubbins we like. Non-linearity comes in many forms in the audio world. The most discussed in digital platforms, is of course the one named Harmonic distortion. Usually people say "distortion", "saturation", "overdrive" and stuff like that to generally refer to harmonic distortion.

We've got plenty of that topic being discussed all around us, and I've got plenty more to add to that... later. For now, let's look at some other types of distortion.


Let's roll with Frequency distortion.


This term covers a lot more ground than just the idea of an EQ curve on your audio. It does include the frequency response across the spectrum, but also includes lots of other things: Transient response, phase delay, rise time, and overall bandwidth (lowest to highest frequency) are all covered by the term. I'm sure there are others... Please feel free to bring anything to the table, I'm still quite green to this myself

So let's expound this non-linear business and see how the total amount of volume gain (or electrical voltage) and the bandwidth of the frequency sepctrum are related in many amplifier circuits. Here is a graph (you knew it was coming!) showing Frequency vs Volume Gain:

Crafting circuit models using plugins-gain-v-frequency-annotated.jpg

(Sorry... no Photoshop handy!)

Now, it's way beyond me to give any kind of actual values here, so please don't go looking for them. I can only show the size and shape of things, so...

The volume gain that an amplifier can provide is often limited depending on frequency. Conversely, frequency response of an amp can be limited depending on the gain it supplies. Keep in mind our axis from the previous graph (x = freq, y = gain) and look at my awfully basic representation of the raw output of a simple amplifier:

Crafting circuit models using plugins-open-loop-gain-v-frequency.jpg

This is a hopefully excuseable attempt to demonstrate the following behaviour in an amp: More gain, less frequency spectrum. Less gain, more frequency spectrum. In a nutshell!

Due to this relationship, you have a situation where if you need more gain from the amp, it's going to cost you some frequency distortion to reach that level of gain. And, likewiase, if you want to reach across a broad frequency spectrum, you may well run out of gain and begin saturating (harmonic distortion!) in order to do so.

Since the job is to amplify, these components have a habit of simply blasting itself to death in very short order with extreme amounts of everything that it does. In doing so, it will probably blow itself up or melt out some other part of the chain.

Amplifiers, by name and nature, can usually extend in either direction in proportions that go far beyond necessary for audio uses. However, they never extend BOTH ways like that. It's a trade-off. So, considering many amp components and circutis are capable of hitting either extreme gain (within a very limited frequency range), or frequencies extending way beyond human perception (providing only a tiny gain boost)... This balance is one huge factor in the design of audio amplification.

We can cut out a section from that graph to find a nice cross-section as such:

Crafting circuit models using plugins-closed-loop-gain-v-frequency.jpg

These white lines represent chosen "operating points" by an amp designer. Elements such as capacitors and resistors have been added to keep the frequency range within a more efficient boundry of useage and the gain controlled. Positive or negative feedback, in which a part of the amp output is fed back to it's input following some more components, was probably also applied. There are various ways to make use of these amplifying parts so that they can give an appreciable function within a music environment. I personally do't know what I'd do with an amp spitting 2 volts of radio frequencies, and even less so an amp giving me 200dB of 50hz.

So amp designers get to work, and find ways to make these components function according to our music-related needs, using some of the lovely gubbins described above.

However, as we live in the real world, these gubbins all come at different little prices of their own. The amp designer is constantly balancing these inefficiencies. Maybe this is why we have ended up with digital? It's probably the way nature fights us back at every turn... much like mixing in digital, amiright?

So I'm gonna move on now to a brief bit about how to measure frequency distortion using square waves, before we start looking closer at what these measurements tell us about what's happening (or what lies) inside the circuit, what to listen for in models of this stuff, and... how to apply some of it yourself using simple, easily available audio tools.

BRB...
Attached Thumbnails
Crafting circuit models using plugins-gain-v-frequency-annotated.jpg   Crafting circuit models using plugins-open-loop-gain-v-frequency.jpg   Crafting circuit models using plugins-closed-loop-gain-v-frequency.jpg  
Old 1st December 2017
  #23
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🎧 5 years
Modelling Frequency Distortion - Square Wave Tests

Now it's mostly just a bunch of pictures, enjoy the ride!



Here is a waveform. You already knew that, obviously. Just saying.

Crafting circuit models using plugins-1khz-sine-wave-6dbfs-sine-tone-generator-.jpg


Let's go for a real stretch?convenient lie here and say "that's what sound looks like" kind of. Well, anyway.

Sound and electricity have a handy thing in common: polarity. They are both periodic, a variation over time. That variation is measured from positive to negative, and back. One trip to negative and back. Sounds like a bad day out.

Crafting circuit models using plugins-1khz-sine-wave-6dbfs-sine-tone-generator-annotated.jpg


The speed of that trip is called frequency. A faster sound wave will be higher in pitch than a slower sound wave. 100 cycles per second is slower than 20000 cycles per second... sorry, yeah you know this. Sorry.

Speakers and mics also use this principle of phase, as you obviously know. Speakers and mics are like A/D converters but actually A/A. So the soundwave gets converted into a voltage, and back into a sound wave. All the voltage stuff is what we are here for, amiright?

If we take that 1 kHz sine and apply odd-order harmonics of equal amplitude ad infitum up the frequency domain, we'd have turned it from a smooth gentle sine wave into a brute block called a square wave:

Crafting circuit models using plugins-1khz-square-wave-6dbfs-sine-tone-generator-airwindows-density-.jpg


Note that the wave is already saturated in harmonics to hell and back, so this reading is not gonna tell us ANYthing much at all about harmonic distortion, since theoretically we can't really apply any more than we already are. Leave that topic for other days,

The square wave doesn't transitio smoothly from positive to negative and back - as you can see, it is simply "fully positive" or "fully negative" - a bit like me at rehearsals - and this ability of amplifier of switching at that speed, is actually partially a function of frequency distortion in general.

Therefore, we can use the square wave through an analogue system to measure frequency response, bandwidth, phase delays, transient response, and other aspects that creep in as the designer applies elements to the amp to control and shape it.

Interestingly, we can also see if this works just as good with some of these much touted plugins I have in the drawer...

I've got some responses I had waiting for exactly this moment... can you believe it?

BRB...
Attached Thumbnails
Crafting circuit models using plugins-1khz-sine-wave-6dbfs-sine-tone-generator-.jpg   Crafting circuit models using plugins-1khz-sine-wave-6dbfs-sine-tone-generator-annotated.jpg   Crafting circuit models using plugins-1khz-square-wave-6dbfs-sine-tone-generator-airwindows-density-.jpg  
Old 1st December 2017
  #24
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Best free audio course ever. This should be fun . . .
Old 1st December 2017
  #25
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🎧 5 years
Frequency Distortion Modelling - Typical Responses

DISCLAIMER: I've gone for very EXAGGERATED examples just so it's nice and clear the actual behaviour here. Try it all for yourself some more if of any interest!


Here's some bandwidth effects on square waves, as a result of narrowing the amplifier's fequency response to get more gain out of it, perhaps using combinations of capacitors and resistors to make an RC Circuit:



Above is a High pass filter, a result of a "DC un-coupling" capacitor. This type of cap is used to remove current from the signal, in order to prevent overloading the amplifier stages and other parts, and sometimes used for other things we might get to later. One thing about DC (direct current) - it's basically a sine wave at 0hz, it isn't periodic like sound or voltage and doesn't play well with audio fidelity for several reasons. Caps are used to control it, and in doing so, they provide a certain amount of high pass enough to block as much 0hz as possible, meaning some amount of low-end roll off (eliminating DC while keeping subs is an interesting consideration...)




Low pass filter, reigning all the radio frequency, the bat and dolphin spectrum, frequency bringing operating limits down to where the human ears actually work (or where it sounds good?).


There are several ways of making the RC Circuit. You can put things in serial or parallel, for example. This will have different effects on the frequency response, by adding resonance and wierd phase shifts, but that aside, the cumulative affect on a square would be essentially like this:




Regarding the phase shift, there will also be some phase shift introduced around the edges of those bandwidths, which again can all vary according to the circuit itself and the parts interacting with it, what you're putting through it, and endless other variables. You might find that the transient response is increased or decreased at different frequencies, depending on the arrangements of all these parts. Perhaps the low end gets a particular thump or grind from the way the circuit is sourcing and sinking current and inverting polarity around 0 Hz, for example....


(Thanks to Airwindows for HighPass, LowPass, and Capacitor used in this example.)
Attached Thumbnails
Crafting circuit models using plugins-1khz-square-wave-high-pass-airwindows-capacitor-.jpg   Crafting circuit models using plugins-1khz-square-wave-low-pass-airwindows-capacitor-.jpg   Crafting circuit models using plugins-1khz-square-wave-low-high-pass-airwindows-capacitor-.jpg   Crafting circuit models using plugins-1khz-square-wave-custom-1-rc-resonance-phase-shift.jpg  
Old 1st December 2017
  #26
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🎧 5 years
Frequency Distortion Modelling - Carrying Voltage Across The Bandwidth

I don't have the best tools or understanding of electrical current to really give the best explanation as to how it can affect audio, I really hope others join in to fill all the gaps and correct my obious noob-ness.



Nonetheless, I gave it a go.

Crafting circuit models using plugins-1khz-square-wave-transient-analysis-reduced-sustain-0hz-700hz-softube-transient-shap.jpg


Here's I've still got that 1 kHz square wave but applied an envelope-shaper style plugin with a multiband function, to suck out all the "sustain" from 700 Hz and below. Notice those cool curves on the tops and bottoms? Things are getting wierd. We like wierd, don't we? Good

So that odd shape is showing you not just the normal slope we saw in a filter like before, but some kind of action going on. As I said, I've sucked out a bunch of low frequency sustain in that one.

Here's another:

Crafting circuit models using plugins-1khz-square-wave-transient-analysis-reduced-sustain-0hz-4000hz-softube-transient-sha.jpg

This example is sucking sustain out from 4000hz up, producing a wierd little nib on the tip of the square.

Amazing, isn't it?

...So what? Right?

Before we do the transients, remember this:


We're seeing how FAST or SLOW the amplifier is.

What we are are seeing is called "rise time" in amp design speak. How fast can the amp rise and fall from positive to negative? As was mentioned earlier, you can't have an infinity amplifier, not for more than 10 seconds before it overheats and explodes on itself. The elements that restrict frequency response, are affecting the rise and fall time of the amplifier.

So - frequency response, frequency means speed of cycle, speed of amp means frequency cycles available to amp.

We are measuring the speed as a function of the frequency response. Again, this is nothing really to do with "distortion" relatng to harmonics et al. we'll get there, don't worry.

How fast can it switch from positive to negative? Will it incur any penalty in doing so? We're looking at the accuracy of an amplifier after all, and that will come alive vividly when we look at transient response.

Next time!
Attached Thumbnails
Crafting circuit models using plugins-1khz-square-wave-transient-analysis-reduced-sustain-0hz-700hz-softube-transient-shap.jpg   Crafting circuit models using plugins-1khz-square-wave-transient-analysis-reduced-sustain-0hz-4000hz-softube-transient-sha.jpg  
Old 1st December 2017
  #27
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🎧 5 years
Frequency Distortion Modelling - Ring and Overshoot

Did you miss me?

Tough, I'm back.


Ok, did you notice on the last pictures, that the low end sucking caused the response to go FURTHER than it should have, when switching between polarities?

And the high-end sucking caused it to reach the correct level, and then drop for a moment before settling to normal again?

Very cool to see we can actually model this stuff in digital to a degree. But how much further does it, and can we, go?

Read on.

Ring and overshoot, my friends. Read 'em and weep.


Crafting circuit models using plugins-1khz-square-wave-overshoot.png

This is Overshoot. Say hi. As your filters in the previous example get a steeper and act more powerfully on the signal, we start seeing that the amplifier tends to go overboard or even backwards once it reaches it's peak. In this example here, we are seeing some quite extreme overshooting at the beginnings of each step. This is a non-linearity of the bandwidth filters used to control the amp, a direct result of the filter slope (think 30dBfs per octave or something) being way too steep and messing with the response. It sounds kinda like a weird "highs-forward" sound, obviously it can also poke nasty holes in your ears with those big nasty spikes if not well balanced, but can also do really cool things to a soundstage or wake up the dynamic activity a bit - it's another thing that designers fight against in their balancing act, and yet we look back so fondly on their enemies, such as transient response overshoot.. and ring.


Crafting circuit models using plugins-1khz-square-wave-ring.png


Ring is another similar degradation of sorts introduced and managed by the filtering. It's osciallting, but not necessarily because of the transient - maybe a steady-state signal with lots of ring would be all wobbly all along the tops and bottoms, without the big overshoots at the beginning... I dunno tbh!!

Just to prove it can be done to great levels of complexity in a DAW environment, and to taste, I modelled both ring and overshoot using two instances of Airwindows's overlooked gem, PhaseNudge:

Crafting circuit models using plugins-1khz-square-wave-overshoot-ring.png
Attached Thumbnails
Crafting circuit models using plugins-1khz-square-wave-overshoot.png   Crafting circuit models using plugins-1khz-square-wave-ring.png   Crafting circuit models using plugins-1khz-square-wave-overshoot-ring.png  
Old 1st December 2017
  #28
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🎧 5 years
And finally!

Just for giggles, consider this the pop quiz section.

I have lots of "analogue emulation" plugins. I put a bunch of favourites through the square wave test. See attachments.

Not a very scientific or measured one, again I stayed at a single frequency of 1 Hz for now. Definitely any of these models will show you all kinds of crazy behaviour variations if you sweep through them and whatnot. And again I went for levels that showed the most interest and detail on the scope. Some don't show much in this case, but that's not a question of good or bad modelling per se. Which leads neatly onto...

Yet another disclaimer: This is not about quality of modelling or talking about which plugins I like/dislike or how much any of this even matters. I'm not gonna do "shootouts" on visuals, that's just silly, and don't generally like that particular sport anyway.

THIS is just recreational and educational. You can deduct what you like from it, I'm sure many will, I definitely don't intend to slander or embiggen anyone who works hard to giev us great tools. Also, some of these results may surprise you now, but then you put them through other tests... the tables turn. We'll get to that another day.

Also I'm not gonna take requests on this. There's a couple of "sure bets" that I didn't post, by choice, because I'm not tryong to stir that particular pot. If you want to indulge in comparisons, even based on my own screenshots, I ask you please make another thread or do it elsewhere - thanks

The fun bit: After all the above useless doodles and verbiage, can you recognize any of those behvaiours in any of these analogue emulating plugins?





Some favourites of mine:

Notice the examples with asymmetrical stuff happening - the Audified is tube-based, as are several of the Soundtoys examples, and asymmetry is regarded as a desirable function tube circuits (though it is neither restricted to them nor are they to it). If you look closely, you might see some with the tiniest of differences on either pole.

Lots of cool overshoot in my favourite-sounding examples. Maybe I'd design a DAW channel strip with a tuneable amount of it.

Don't deny it, those AO eualizer plugins have a lot of sound under the hood. In this test that certainly seems to be the case. The Combox showing strong overshoot, probably a sign of decent work modelling the saturating inductors in the EQ section (overshoot is common in magnetic components). Any surprise that this baby seriously wakes those dynamics up? LCF showing massive overshoot, definitely lots of inductor/transformer modelling going on there!

The Slate stuff doesn't show much here (notice the preamp and the EQ actually are a bit different!). Nice assymetry. Their saturation models measure extremely well in tiny detailed ways in other tests (next time) so I trust Fabrice/Steven that this is fairly accurate for 1 Hz at "whatever" input level, sure I have no reason to doubt it.

What about those Airwindows ones? Those are some very powerful algorithms in there, no doubt. I love one thing with this one in use particularly - when you push the input slider up, the response is the same but louder. When you push the output slider up, you change and morph the frequency distortion with it. So it must have some post-output algorithm going on, which I always find very preferable personally over a redundant gain knob. Just me , I know.

Especially interesting also is TapeDust - that one is adding some kind of tapey wow/flutter to the signal, as can be seen by the blurring. More on this in other tests, but cool to see here.


And what about all those Neve modules? None of them seem to match at all. The Burnley doing serious HF limiting, probably some phase delay too. The Slate's are assymetrical, which actually is very reasonable for a transistor circuit (more on that particular quirk later). Similarly the API's too are quite varied there.

Which one is most accurate? Guess we'd need to measure the real thing to know...


If anybody wants to add, subtract, or multiply any of the above, I'd enjoy hearing from you!

Meanwhile I've got a few nice new saturators to demo for next time.

Peace.
Attached Thumbnails
Crafting circuit models using plugins-1khz-square-wave-black-rooster-audio-v73pre.jpg   Crafting circuit models using plugins-1khz-square-wave-soundtoys-decapitator-n-.jpg   Crafting circuit models using plugins-1khz-square-wave-soundtoys-little-radiator-bias-.jpg   Crafting circuit models using plugins-1khz-square-wave-soundtoys-radiator-mic-.jpg   Crafting circuit models using plugins-1khz-square-wave-soundtoys-sieq.jpg  

Crafting circuit models using plugins-1khz-square-wave-kush-omega-.jpg   Crafting circuit models using plugins-1khz-square-wave-kush-omega-n.jpg   Crafting circuit models using plugins-1khz-square-wave-sly-fi-axiseq.jpg   Crafting circuit models using plugins-1khz-square-wave-slate-digital-fg-73.jpg   Crafting circuit models using plugins-1khz-square-wave-slate-digital-fg-n.jpg  

Crafting circuit models using plugins-1khz-square-wave-analog-obsession-combox.jpg   Crafting circuit models using plugins-1khz-square-wave-analog-obsession-lcf.jpg   Crafting circuit models using plugins-1khz-square-wave-airwindows-busscolors4-lush-.jpg   Crafting circuit models using plugins-1khz-square-wave-airwindows-busscolors4-punch-.jpg   Crafting circuit models using plugins-1khz-square-wave-airwindows-busscolors4-rock-.jpg  

Crafting circuit models using plugins-1khz-square-wave-airwindows-busscolors4-steel-.jpg   Crafting circuit models using plugins-1khz-square-wave-airwindows-tapedust.jpg  
Old 1st December 2017 | Show parent
  #29
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🎧 5 years
Quote:
Originally Posted by wjmwpg ➡️
Best free audio course ever. This should be fun . . .
Hehe, with no audio, of course! But we'll get there.

I should certainly mention the tool I used to learn all this and do the measurements, also for "designing" stuff within the DAW:

J-Scope by Jagged Planet

J-Scope VST Oscilloscope

Sorry Mac folks, but oscilliscopes aren't hard to find these days.

Oh, and the other tool is Google.
Old 1st December 2017
  #30
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🎧 5 years
Interesting stuff..keep it coming..
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