Let's talk about reverb, will you?

[Goobers]

 

Again what? Again, hardware devices tend to smoke plugs because they are designed for limited purpose and that design and construction goes into optimizing the device for a given purpose. I explained that as clearly as it can be explained. It's now up to the reader to gain the basic knowledge required to grasp it... the concepts, the vocabulary... basically how things work. I know how things work. I

[Lee Flier]

But how do you know that's the plug? The plug by definition has to reside within a DAW, and you'll find plenty of people who say that DAWs don't have the same depth/dimensionality of analog gear, even if you're just feeding in audio. For all you know the plugs may be perfect, but limited by their environment.

 

Yeah, I have definitely noticed this. The biggest issue with mixing ITB is, IMO, reverb tails (whether natural or artificial reverb). If I mix on my analog console and I'm using any reverb plugs, just taking taking the output from the plugin and sending it to 2 channels in the console gives quite a bit more depth and dimension to even cheap reverb plugins. It's not as noticeable with Reaper as it is with Pro Tools , but it's still noticeable. And I HAVE done numerous blind tests where I can distinguish an ITB from an analog console mix, usually because of the reverb tails. Admittedly it's been a few years since I've done such a blind test, so there may have been some improvement in recent years, but I do still notice on a casual basis that sending the verb out to a console still improves the depth and "width" of reverb, even if it's a plugin. So it's entirely possible that the "problem" with reverb plugs isn't with the plug per se, but if you don't want to go out to an analog board, hardware reverb may have an advantage.

 

One thing I don't think I've tried yet is the UAD2 reverb going out to analog vs. ITB. I'll have to try a direct comparison there when I get a chance.

[JeffLearman]

 

So Lee, what do you think- same algorithm, different processor- same outcome, or different? We have Lexicon making the PCM96 hardware and the PCM Native bundle, with the same algorithms... do you think the algorithms lead to different results depending on the processor? I don't think that's a subjective question. Do you? Do we need to have discussions about the "sound" of different processors?

If it's the SAME CODE running on the SAME DSP, then no, there's no difference. If it's a software emulation of the DSP, then no difference. But if it's the same *algorithm* but *coded differently, for different hardware, with different bit depth*, then the answer is "no, it doesn't necessarily sound the same".

 

Most reverbs have 24-bit fixed-width DSPs, using 48-bit fixed-width for intermediates. Computers use 32-bit float, with 64-bit float for intermediates. Which would tend to make the computer sound better, in theory -- unless the quantization noise somehow improves the results.

 

It's like saying "would a numerical algorithm produce different results on different computers?" and the answer is a resounding YES, if the algorithm is subject to range and precision issues, and range and precision differ on the two computers. (IEEE floating point was introduced specifically to help minimize these issues.)

 

However, how much it differs depending on the type of algorithm. For example, a simple delay is the least effected, since it's the simplest algorithm, and merely additive (with scalilng). Reverbs are a combination of delays and EQ. (Convolution reverbs are only delays, so convolution-based reverbs would be almost completely unaffected by range & precision issues, except in the low level noise band and when clipping. Oddly enough they sound like EQ, but ... they aren't, not really. They're just lots of additive delays.)

 

But I don't know enough about other kinds of reverb algorithms and can't comment on their susceptibility to differences in bit widths.

[JeffLearman]

I've always thought that when you use a reverb program as a plugin in a DAW as opposed to a reverb program in a hardware box, it's having to interact with a lot more stuff in the computer than it would have to in the hardware version. The operating system, the recording program, drivers, other plugins and programs are all taking up resources that the hardware version doesn't have to deal with by itself. The hardware version can go from point A to point B without interference in a straight line. I have also read or heard that one reason hardware boxes supposedly sound better than plugins is because the processor chips in hardware boxes are more efficient for the tasks at hand. (In my mind I always thought of that as meaning they go from point A to point B in a straighter line) I've seen advertisements over the years from companies that tout the type of processors used in their products. For example Peavey states that their new amps use SHARC processors for their effects. I have a friend who has one of these amps and I really like the effects. Especially the reverb. Why would Peavey advertise the processing chip used in it's effects if it didn't have anything to do with sound quality? In my mind it's because the chip is probably optimised for digital signal processing. It probably gets from point A to point B in a straighter line than other chips that are not optimised for DSP tasks. Or maybe I'm just falling for an advertising gimmick.

Interesting theory, but your guesses turn out to be mostly wrong.

The plugin itself is fed a stream of data and it's told the sampling frequency. All it has to do is take a bucket of numbers (a "buffer") and compute the corresponding output bucket of numbers. (Of course, it often needs to remember some info from previous buckets.) All that other computer OS and hardware stuff is factored out by the design of the plugin interface. Those things are for the DAW to deal with.

Conversely, DSPs used to have to deal with the fact that they're handling data in real-time, and the coder had to carefully count the number of instructions on one path and make sure it exactly matched the number of instructions on another path, or stuff would get out of phase. That's not the case with modern DSPs, which also used buffered data.

The efficiency of a processor for a given task only affects how complex its algorithms can be, before it runs out of gas. The only things that can affect the tone are the bit width and the code.

I've been doing dedicated real-time programming since the late 70's. Not much audio, but I understand the principles, and I've studied Shannon's book "Digitial Signal Processing", which is the cornerstone. I admit that some of the math in that book is a bit over my head. But I do understand how math in general works in computer processors.

[JeffLearman]

I'm curious to hear what Bech has to teach us about the differences between hardware and software designs.

 

I did a little VST coding, and understand that interface. I've only done a little study of DSP processors, most of that way before the modern era. (I even wrote code to turn a processor intended to be a DSP into something like a network router, back before Arpanet became the Internet.)

 

If the DAW is operating properly, audio latency between input and plugin, and between plugin and audio output, is constant. That should also be the case for a hardware reverb. Granted, there's a lot more fuss getting a general purpose computer to the point where you have low latency without dropouts. But once you get there, and as long as you have no dropouts, everything else is just the plugin code making the plugin sound the way it does.

 

Note that I'm not addressing things like Lee's assertion that reverbs sound better after being fed through analog circuitry. Anything could sound better after going through any kind of processor, whether analog or digital. Right now I'm only talking about the difference between using a plugin reverb, versus using S/PDIF to a hardware reverb. The only differences would be differences in the code and the processors. IMHO, that would be vanishingly small in the case of the Lexicon 96 versus the plugin, based on what the coder said about the port. (It would be even smaller, if the 96 uses a floating point DSP with IEEE floating point.)

[JeffLearman]

Correction: I see that the 96 runs floating point, presumably IEEE, written in C++ just as a VSTI would, so porting the numerical algorithms shouldn't have any effect on the sound.

[Lee Flier]

 

If the DAW is operating properly, audio latency between input and plugin, and between plugin and audio output, is constant. That should also be the case for a hardware reverb. Granted, there's a lot more fuss getting a general purpose computer to the point where you have low latency without dropouts. But once you get there, and as long as you have no dropouts, everything else is just the plugin code making the plugin sound the way it does.

 

 

It should be, yes. I'm not entirely convinced that it is. I think there are still minute timing errors between tracks in DAWs once you start adding DSP (which happens pretty much right away unless you're leaving everything at unity in the DAW and going out to an analog console from there). This may be getting to be less and less an issue as the years go by, but I think it's still there. I admit I can't prove this, and that while I'm a programmer, I'm not quite geeky enough to be bothered to start delving into that kind of minutia. But I do know that my ears can tell there's a difference, and I'm not at all sure it's just the difference between the plugin and whatever the analog circuitry adds.

 

 

Note that I'm not addressing things like Lee's assertion that reverbs sound better after being fed through analog circuitry. Anything could sound better after going through any kind of processor, whether analog or digital. Right now I'm only talking about the difference between using a plugin reverb, versus using S/PDIF to a hardware reverb. The only differences would be differences in the code and the processors. IMHO, that would be vanishingly small in the case of the Lexicon 96 versus the plugin, based on what the coder said about the port. (It would be even smaller, if the 96 uses a floating point DSP with IEEE floating point.)

 

 

Yeah, in theory there wouldn't be any difference at all. I haven't done that direct comparison with Lexicon hardware vs. plugs, so I couldn't comment on that. But it wouldn't surprise me either way (if there was a difference or if there wasn't).

[JeffLearman]

It should be, yes. I'm not entirely convinced that it is. I think there are still minute timing errors between tracks in DAWs once you start adding DSP (which happens pretty much right away unless you're leaving everything at unity in the DAW and going out to an analog console from there). This may be getting to be less and less an issue as the years go by, but I think it's still there. I admit I can't prove this, and that while I'm a programmer, I'm not quite geeky enough to be bothered to start delving into that kind of minutia.

It's an easily tested hypothesis, simply feeding in a signal that has an obvious visual marker, do a mixdown, and look at the results lined up with the original. I've done stuff like this, and had a couple surprises, mostly with plugins that had latency that wasn't properly compensated for. I've also seen the default 44.1/48 kHz conversion that builtin soundcards do have really strange effects on timing and synchronization.

[UstadKhanAli]

 

Okay, so for those of you who like using OTB hardware, particularly those who use/have used mono input(pedals), do you simply bus out L & R from the mix to hardware insert, and track each side mono onto a new instrument track?

 

 

I'm using a TC Electronic M300 hardware reverb via S/PDIF. That'd be stereo. In my DAW, I simply set up a stereo auxiliary track, which outputs to the hardware reverb, and then send a signal via a send from whatever tracks I wish to have reverb to the auxiliary track, just as I would with a console. I control the amount of reverb that gets sent to the auxiliary track by each send.

 

 

Second question: Hardware guys, how hard are you hitting the rack verb inputs? All trial and error, or do you have a favourite starting point, regardless of song vibe/reverb type?

 

 

I'm hitting it as hard as it needs to be hit, but not going over, pretty much using common sense the way one would when using a console and going to the analog I/O of a hardware reverb or a plugin. Good healthy amount, but nothing that starts distorting.

[Lee Flier]

 

It's an easily tested hypothesis, simply feeding in a signal that has an obvious visual marker, do a mixdown, and look at the results lined up with the original.

 

 

But there will always be differences introduced by the analog circuitry, so it becomes impossible to distinguish between that and any shortcomings in the digital processing.

 

Obviously if I had both the Lexicon hardware and plugin, it would be easy to test that comparison (via SPDIF connection, anyway), but I don't.

 

 

I've done stuff like this, and had a couple surprises, mostly with plugins that had latency that wasn't properly compensated for.

 

 

Yup... definitely I think PDC isn't 100% perfect. I think there may still be similar issues with gain changes, too.

[Rekel]

Yup... definitely I think PDC isn't 100% perfect. I think there may still be similar issues with gain changes, too.

 

 

PDC in Nuendo works just fine over here.

[Folder]

Interesting theory, but your guesses turn out to be mostly wrong. The plugin itself is fed a stream of data and it's told the sampling frequency. All it has to do is take a bucket of numbers (a "buffer") and compute the corresponding output bucket of numbers. (Of course, it often needs to remember some info from previous buckets.) All that other computer OS and hardware stuff is factored out by the design of the plugin interface. Those things are for the DAW to deal with. Conversely, DSPs used to have to deal with the fact that they're handling data in real-time, and the coder had to carefully count the number of instructions on one path and make sure it exactly matched the number of instructions on another path, or stuff would get out of phase. That's not the case with modern DSPs, which also used buffered data. The efficiency of a processor for a given task only affects how complex its algorithms can be, before it runs out of gas. The only things that can affect the tone are the bit width and the code. I've been doing dedicated real-time programming since the late 70's. Not much audio, but I understand the principles, and I've studied Shannon's book "Digitial Signal Processing", which is the cornerstone. I admit that some of the math in that book is a bit over my head. But I do understand how math in general works in computer processors.

I've never heard of Shannon's book "Digitial Signal Processing" and I'm terrible at math so please bear with me.

I understand the principle of buffering, but processing chips have different architectures. Some are "supposedly " better at some things while others are "supposedly" better at other things.

If all chips are just crunching numbers and the results are all the same then why do companies tout the type of chips used in their products?

About fifteen years ago I bought a "Lexicon Alex" effects unit because the ads for it touted that it used their patented LexiChip. I though Lexicon is known for great reverb so if it has a LexiChip in it it must sound great.

It was probably the worst $300 I've ever spent in my life.

On the other hand the new Peavey guitar amps tout that their effects are run on SHARC DSP processors. These amps have probably the best digital reverb I've ever heard in a guitar amp.

So in my non-mathmatical mind I think LexChips suck and SHARCs sound fantastic.

I know that the software algorithm is probably the most important aspect, but why couldn't the chip contribute to sound quality as well?

Where am I going wrong here?

[UstadKhanAli]

Just as there are a lot of bonuses to plugins.

 

Yes, definitely.

 

I can use it independently on any number of tracks, up to what my CPU can handle.

 

Yup, having different independent settings is definitely a bonus. That said, how many reverbs are you going to use on a mix? I do occasionally use a second reverb on a mix, in which case I'll either use a plugin or simply print a different reverb from the hardware reverb, but I don't remember the last time I used more than two reverbs on a mix, and often only use one. Or none. Maybe you use more independent reverbs than me, in which case this becomes a bigger advantage.

 

I can manipulate it during mixdown in real time if I wish to do so.

 

Cool. You can automate it to if you wish, which can be a major bonus. It's just really easy to grab the knobs on a hardware reverb and start going, and for me is far better than manipulating something with a mouse. As always, YMMV.

 

I can do an offline mixdown, much faster than a real-time mixdown.

 

Yup. That's definitely an advantage.

 

I never have to worry about it getting zapped or failing due to hardware component issues.

 

I've been using hardware reverbs for a kajillion years and I've never had this happen. I still use my original Quadraverb. Now, if you compare that to fussing/maintenance with plugins, I'll take a hardware unit any day. Besides, your typical computer is far more susceptible to crashes, weirdness, getting zapped, failing due to hardware component issues, etc. than a hardware reverb, and has far more moving parts. Sorry, I don't see this as an advantage of plugin reverbs at all.

 

There are no ground loops to worry about, no outboard latency issues to worry about, and no cables to fuss with.

 

Do you get ground loops when connecting a hardware reverb via S/PDIF? I've never heard of such a thing, but don't know for a fact.

 

Outboard latency: I don't know a single person ever in the history of mankind who was ever worried about outboard latency on a reverb hardware unit. Do you feel that 2-3 milliseconds would adversely affect your reverb?

 

Cables to fuss with? It's S/PDIF. Connect. Connect. Bang. Done. Just took half a minute. That's a helluva lot faster than the time it takes to install a plugin. And typically, you get to install a plugin several times if you upgrade stuff. No, no advantage here either.

[aZnrockstar]

 

I'm using a TC Electronic M300 hardware reverb via S/PDIF. That'd be stereo. In my DAW, I simply set up a stereo auxiliary track, which outputs to the hardware reverb, and then send a signal via a send from whatever tracks I wish to have reverb to the auxiliary track, just as I would with a console. I control the amount of reverb that gets sent to the auxiliary track by each send.

 

 

Yeah, that makes sense. I'll have to try it.

[Anderton]

It's not an opinion question, it's a fact question. Either the algorithm is the same or it isn't.

 

Goobers, goobers...don't you know that the pre-CBS "16A55C547FFC" has a much warmer, rounder sound that the usual"16A55C547FFC" we have to put up with these days?

[Anderton]

 

It should be, yes. I'm not entirely convinced that it is. I think there are still minute timing errors between tracks in DAWs once you start adding DSP (which happens pretty much right away unless you're leaving everything at unity in the DAW and going out to an analog console from there). This may be getting to be less and less an issue as the years go by, but I think it's still there.

 

 

I've actually delved into this, and both your observations are mostly correct. Some programs are extremely good about doing path compensation, to the extent where if you throw tracks out of phase, they cancel completely (as in, inaudible). But with others, you get that "offset by a couple samples" sound we all know and, uh, don't really love very much.

 

I think it may also depend on the plugs, IOW, certain plugs seem to be more accommodating to compensation than others...but I haven't gone THAT far to quantify it.

[Anderton]

I've also seen the default 44.1/48 kHz conversion that builtin soundcards do have really strange effects on timing and synchronization.

 

If you're talking Windows, there has been a long-standing bug with sample rate conversion that's finally fixed with a hotfix.

 

to Garrett Haines for turning me on to this.

[Anderton]

 

Yeah, that makes sense. I'll have to try it.

 

 

Using S/PDIF is definitely the way to go, if your hardware has it...you avoid those extra stages of conversion.

[JeffLearman]

But there will always be differences introduced by the analog circuitry, so it becomes impossible to distinguish between that and any shortcomings in the digital processing.

True, unless you're using S/PDIF.

 

Even "in the box", there can be timing discrepancies. Not when both plugin and DAW are working right, but in practice, there are a few fiddly bits that are easy to get wrong, both in the DAW and in the plugin. A plugin is allowed to add latency (some effects are impossible without it, such as "pre-echo".) It's supposed to report its latency, but might not do so correctly. The DAW is supposed to compensate for that, but it gets very tricky to do this correctly when routing is flexible -- and I suspect, impossible in the case where routing is flexible enough to allow feedback loops.

 

For years I used a DAW that did plugin latency compensation, only to read a bug report one day that explained that it didn't do it correctly in certain cases.

[JeffLearman]

That said, how many reverbs are you going to use on a mix?

It's a bigger bonus for compressors, I'll warrant. But as I said above, I'll use quite a few reverbs even in a fairly simple mix, just for the early reflections mentioned above. Say, three acoustic instruments, like guitar, fiddle, and mando, in that order left to right across the soundstage. On guitar, I want early reflections fast on left and slow on right, so that's 2. For center, I want early reflections almost (but not quite) the same, so that's 2 more. Mando? 2 more, this time with the faster reflections on the right, slower on the left. Then a stereo reverb on the stereo mix, for a total of 8 channels (4 stereo channels).

 

It's just really easy to grab the knobs on a hardware reverb and start going, and for me is far better than manipulating something with a mouse.

No doubt. The only advantage to the mouse is that the controls can easily change with the application.

Do you get ground loops when connecting a hardware reverb via S/PDIF? I've never heard of such a thing, but don't know for a fact.

You can get ground loops with anything you can plug into the wall. Lately I'm having problems with ground loops between my MOTU 828, my laptop, and my mixer. arrgh.

 

Outboard latency: I don't know a single person ever in the history of mankind who was ever worried about outboard latency on a reverb hardware unit. Do you feel that 2-3 milliseconds would adversely affect your reverb?

Right, not a big issue for reverbs. A bigger issue for other plugins.

 

Cables to fuss with? It's S/PDIF. Connect. Connect. Bang. Done. Just took half a minute. That's a helluva lot faster than the time it takes to install a plugin. And typically, you get to install a plugin several times if you upgrade stuff. No, no advantage here either.

It only takes me a couple clicks to add a plugin to a track, and it doesn't use up ports on my computer keeping me from using other plugins.

 

I'm not saying plugins are better. They are really damn convenient, though, and bang for the buck is high.

[JeffLearman]

If you're talking Windows, there has been a long-standing bug with sample rate conversion that's finally fixed with a hotfix . to Garrett Haines for turning me on to this.

My understanding was that the Soundblaster itself (or whatever) was doing the conversion. In any case, it dates back to early 2000's, so isn't any problem that would be new with Vista. And it happened when using ASIO4ALL with builtins.

 

The problem as I understood it was that a lot of builtins and soundcards like the mighty Soundblaster were built to run in 48K mode, and if you feed them 44.1, they did an internal conversion on the fly. (Maybe the OS was involved, but I got the impression otherwise, though don't recall the details.) Anyway, it wasn't a very good job of on-the-fly rate conversion.

 

This caused severe problems for internet collaborators. Someone would post a track, someone else would load it, record their part, and post a quick mix. When the quick mix was lined up in the original poster's DAW alongside the original track, and adjusted to align at the start, the synchronization would drift off. I practically went blue in the face trying to explain to people that this could NOT be caused by different clock rates in different computers. (That would cause a pitch discrepancy -- one too slight for anyone but Superman to hear.) Eventually someone posted the cause, and the easy solution (e.g., convert rates using r8brain or whatever, when using a built-in), and the problem was a thing of the past.

 

All of which has nothing to do with reverbs. Maybe that problem was really due to software, and maybe it predated Vista, and is solved -- that would be nice.

 

BTW, there are telltale ripples in the RightMark results for interfaces that do this conversion. Let me know if you'd like to see, I think I still have that stuff on line.

[Beck]

Plugs don't have the analog components of the hardware versions, which of course influence the sound. Whether that sound is desirable or not is in the ear of the beholder, and what effects (if any) you put before or after the plug-in that are different from the hardware unit. For example, if you were to put one of the URS "transformer emulation" plugs in front of a digital reverb, you might like the resulting sound better than the analog version OR the digital version Those "vintage digital" hardware units often used custom chips, and chips like 1st and 2nd gen Motorola DSP, that were powerful at the time (certainly, you were much better off designing an audio card around a Motorola 56000 DSP than a 68000 CPU). However, today's multicore processors outperform those "vintage" DSP chips by several orders of magnitude. I don't believe soft synths like the FXpansion Synth Squad or the u-he Diva could have been implemented economically with the hardware DSP chips that were available even ten years ago.

 

While that

[Anderton]

Did the PCM70 sound good to you when it came out? It did to me and to countless others around the world that used it on recording projects with fully professional results.

 

BTW first of all, I said PCM80 above as the unit where the reverb was shaky as opposed to PCM70. Sorry about that.

 

Anyway, I (still) have a PCM70 and honestly, I wasn't that knocked out by the sound. I never really used it much, I could get much better reverbs with other units...the multivoice chorus stuff sounded okay, though. Anyone want to buy it?

 

As to "If proprietary ICs, say for example like those in the venerable Lexicon PCM70, are more than adequate for the task at hand, anything more is overkill, a waste of time and money, and adds nothing to the performance of the device" I think that may be missing my point...which is that those old dedicated chips weren't very powerful, so today's chips can indeed run a computer, do housekeeping, and still have more horsepower left over than the dedicated chips of yore.

 

Ultimately, I agree that one must choose what sounds best. Which, frankly, is why my hardware boxes are gathering dust, and I'm building up a really nice collection of effects chain presets for reverb using various plug-ins. As for vintage reverb, I really think anyone would have a hard time differentiating between a hardware 224 (Lexicon) and a hardware 224 (Universal Audio); when I want that 80s vibe, the UA 224 works just fine.

[Beck]

Well the point is that a lot of people did like the PCM70 and other hardware verbs. So I could have pulled out any example. And even though I like the PCM70 for echo and verbs I like the LXP-15II even better and the that's the one of the two I ended up keeping.

So back to the point about the proprietary chips... they were as powerful as they needed to be. And keep in mind, even though I prefer a lot of the older outboard hardware processors, new hardware verbs are still made today with even more powerful processors, if that's important to you, and those too have all the advantages of a dedicated device compared to a plug. We're talking reverb and delay/echo here, at least I still am, so I don't want or need a processor that gives me the spatial effects I need plus washes my car and takes my dog to the park to play frisbee. So again, while a powerful processor has a certain gee whiz effect it doesn't really do anything for me. It doesn't move me, and it doesn't necessarily mean a thing in the real world of recording.

I've always chosen my tools very carefully and most of them were ahead of their time... and still are. Whether you use an older or newer hardware processor you have the advantages of a dedicated device, and those advantages have never changed. And the older processors I do have still work as they always have. I can still go back and read the reviews of some of the devices I still own, quite a few written by you. The reverbs and delays sounded fantastic then. How did they suddenly become weak in inadequate all of a sudden? There are a couple reasons it could happen. One is that human hearing has evolved over the last 20 years. Well, that didn't happen. The other reason is that manufacturers and vendors have to keep hocking new products and all those old products are outdated (whether they really are or not) because people have to eat and pay the rent. I think most of us know the latter is what's driving the market. It sure isn't quality. Quality stopped being the point of reference long ago, so a better processor on paper doesn't necessarily translate into a better performing device or plug as far as reverb is concerned.

And that's why people are still falling all over themselves to find some of their fav vintage products that they dumped prematurely. I've done it myself... got rid of an older device to by the new model only to learn the hard way that despite the more powerful processor and better specs on paper, the sequel just didn't cut it. In fact, I've owned two Lexicon LXP-15II's in my lifetime; the one I had for a few years and sold to buy something newer, and the one I have now that took me two years to find back after I realized my mistake. And plugs? Egads! I've tried a great many over the years and I had high expectations for the more recent Lexicons, but they aren't anywhere near as lush and deep as those on my LXP-15, and as you know the LXP is more in the budget level of Lexicons boxes, but it's well designed and you wouldn't know it's not in their top of the line by listening. Glad I have it back. And sure, as I said from the start, I'll use plugs in combination with outboard effects, but I haven't found a plug yet that I'd use alone.

[roomjello]

But how do you know that's the plug? The plug by definition has to reside within a DAW, and you'll find plenty of people who say that DAWs don't have the same depth/dimensionality of analog gear, even if you're just feeding in audio. For all you know the plugs may be perfect, but limited by their environment...just like how the older digital hardware units were limited by having analog front ends and back ends with 12- or 16-bit converters. Then again maybe that was part of the charm of their sound that the plugs don't duplicate.

 

 

 

The Hardware maintains its superiority in the same environment.

So the plugs must be flawed in creation comparatively.