Just a thought that hit my mind while reading the thread about digital delay chips that were available at RS in the past:
Quote from: Mark HammerThe price difference [between bucket brigade and digital IC's] is precisely why I encourage people to set aside their analog preoccupations and go digital when making delays these days.
I would be willing to argue that the terms "analog" and "digital" with reference to delay are not as meaningful as the difference between analog and digitally accomplished distortion, tremolo, compression, etc.
the reason I say this is because bucket brigade delay chips (the heart of "analog" delay) require sampling, which produces many of the potential aliasing and sampling degradation problems of digital systems (in fact, probably to a greater extent). People talk about the "warm" low-pass sound of analog delays, but this can be done digitally just as well. It's how you process the delayed signal in this case. I am more prone to side with Mr. Hammer on this one about encouraging people to go digital in building delays.
"Analog Delay" used as a mark of quality is more of a marketing mojo word than an accurate descriptor of reality.
If you want a truly analog delay, build a tape delay. They have problems too, but at least it IS analog, in that the signal is not sampled. I think the imperfections of the tape delay is what makes it attractive, actually...
However, there is a rational defense for those who wish to build analog delays and choruses for nostalgic or aesthetic purposes. Good art is worth high cost.
Yes, I for sure agree with you. My personal view is that most effects are basically "made" to be done analog-and I see no reason to go to digital for any of it except for the attractiveness of patches and presets. Quite simply though, it's done well analog, and it always has been, so why change it? If it ain't broke, then why fix it?
For analog delays, I'd say the more "lo-fi" (particularly the PT2399) are equal in sound to their analog equivalents. DSP isn't so much fun.
-Colin
Quote from: TransmogrifoxPeople talk about the "warm" low-pass sound of analog delays, but this can be done digitally just as well. It's how you process the delayed signal in this case.
Theoretically yes, but the bottom line is no stompbox delay I've ever tried sounds like my KMD Analog delay. Period. I like the grungey charm of my digital PDS 8000, but it's just not the same. The self-oscillation and ability to twist the delay time knob for pitch change are features usually lacking in a DD (although the PDS 8000 has the latter). I look forward to either a DD design that actually sounds and performs like my analog delay, or for quality BB chips to once again become cheap and plentiful.
Kerry M
Quote from: bwanasonicability to twist the delay time knob for pitch change are features usually lacking in a DD
Kerry M
I suspect that any simple DSP based pedal would be amenable to having the clock wobbled! Any radio hams here want to fire up the RF generator & frighten a FX?
I have BOSS DD-2 and it is the most analogue-sounding DD I've ever heard. It sounds warm and nice and it has the long delay times that a digital delay usually has. I'm pleased with it and I use it a lot!
A heee-YOOOOOJ chunk of what people attribute to analog vs digital delays has to do with the filtering used (or not used).
Imagine a fabulous digital delay with, oh, say 32-bit resolution, and 4k of 32-bit RAM. You wouldn't get much delay, but what you got would sound absolutely pristine and a faithful copy of the original input signal.
In the case of analog delays, few will have more than the equivalent of either 4k or 8k or storage space, which means that generation of any appreciable delay time requires slowing down the signal's progress through the stages. Of course, being analog, each sample has essentially infinite resoution. While that infinite resolution helps to provide a more accurate reflection of the signal at each moment in time, the requirement to stall a bit on how often you take samples degrades the fidelity of the sampled signal in time. It,s a bit like having a 5 megapixel digital camera and running it in movie mode a 4 frames per second. Sure, each picture is crisp, but the movement is jerky and unrealistic.
For analog delays, you can squeeze more time if you slow things down, but whatever you gained in terms of sample resolution is lost in terms of resolution over time. That second type of loss shows up in poor high frequency response and a certain graininess and grittiness. It is also important to note that the longer you hold any sample in a BBD, the more leakage of that signal you experience since it is simply sitting in teeny-weeny capacitors just waiting to bleed signal if you let it sit too long. So, drawing out the delay time too long in the analog domain not only loses you accuracy over time, but also loses accuracy for each sample as well.
The disaster that digital delays court (though not nearly as much as they used to) is that trying to represent an analog signal with insufficient resolution (compared to what the human ear can detect), creates audio impurities n the form of aliasing noise/harmonics. If you sample fast enough, and have enough resoution in each sample, all of that shifts up high enough in the spectrum that it is essentially non-existent for human purposes. (Though I'm sure some of the "golden ears" types will talk about how 128khz distortion affects this or that).
What delays of EVERY kind do fix these dilemmas is filter, filter, filter. Because there are essentially no limits to how much RAM can be used to accomplish any given delay interval (where there ARE limits to how many stages of analog delay you can cascade before what's in the "bucket" looks only marginally like what started out), digital devices can get over some of the hurdles better than analog ones can. Back when anything more than 8-bit resolution automatically put you in the "pro-audio" pricetag category, BBDs competed favourably in terms of quality per dollar spent. While the 10-bit resolution of the Princeton and Holtek chips is not in Lexicon territory yet, 10-bit plus some filtering sounds as decent as analog with the same amount of filtering to address signal loss and undersampling-related problems.
Certainly one of the things that rarely gets talked about is the difference between how the devices are designed and what it is in nature that they are used to emulate. REAL echo and reverb has changes in harmonic content over time, depending on the physical space it occurs in. In contrast, a great many delay units aimed for the most bandwidth they could provide at any delay time, and that's how people used them (although some, like the MXR/Ross analog delay and the MXR digital delay used tracking filters, their intent was to provide maximum bandwidth at any delay and not to deliberately simulate the warmth of long echoes).
As it happened, because analog delays needed to have heavier-duty filtering to address their flaws than digital units did, they more closely approximated what happens with echoes in nature. Both analog and digital were simply trying to keep audio garbage out of the way, but what analog HAD to do inadvertently came closer to "the real thing" than what digital HAD to do.
Of course, what they HAD to do is separate from what they COULD do. Which brings us to Scott Swartz's delay projects that apply the same kinds of filtering to both analog and digital delay technologies, and come up smelling and sounding identical.
The filtering is dirt cheap, and now so are the delay chips. Absolutely no reason to HAVE to use analog for longer delays. On the other hand, where short delays are needed for swept effects (chorus, vibrato, flanging), BBDs make design a whole lot easier and cheaper, so there is every reason to keep using them in those applications.
I agree...even the creator of the pt80 and the bl3208 one (first,digital,second analog)said he had a hard time finding sound differences between them...the most important factor of analog felay sound is the filtering,and i think the compander has something to do,also...but weights less in the final result..regarding the mentioned pds8000,it can oscillate if the trimpot is calibrated to do so...
I received a vintage memory man as a present from a friend,the 4 sad1024 one,and did not find adavntages over my pt80,with the ldr-lfo mod for chorus vibrato...maybe i find a better use for those 4 chips,maybe an ada flanger clone...
thanks Mr. Hammer,for your article...it's now on my collection...
Maneco
Analog vs Digital delay.
At a first listen, you may not notice much difference between
a very nicely done digital delay and an analog one.
but, after a while, you will no longer be able to ignore the digital artifacts, and it'll bug you. Keep adding LPF to mask the artifacts and you end up with the same problem POD guys have- dead, dry, lifeless tone.
Analog delay rules. Always will.
As long as someone keeps making them, that is....
Ed R.
I must say, some of my songs use that delay and in one song I keep playing a low dropped D while playing a short riff over the top... It sounds like it's oscillating at some point:D which does bug me a bit... :?
I still side with Mark Hammer: With proper resolution, sampling rate and memory usage, one can filter to tailor to the "analog" delay sound. I know the frequency change while changing delay speed is not unique only to analog. I have actually programmed a patch on my RP20 that does this at the tap of a foot. Sounds very nice. I have even put my entire RP20 into an analog feedback loop with the "dry" feed through turned all the way down so only the delayed signal comes through. Low pass filter it, and a very "analog" sound comes out...
So even though many commercially made digital pedals fall short of the mark, it IS possible to do it digitally and inexpensively.
Ed,
I expect, and *re*spect, that everyone is likely to encounter something a little different in terms of the delays of each type they've worked with, and the circumstances under which they needed to use them. So, it may well be that one person finds analog delays consistently work better for them, another finds that they come out about even, another finds digital does what they want best, and another finds it varies from model to model and occasion to occasion.
For me, the real questions these variations prompt are:
1) what exactly was it in the sound that led you (or anyone) to prefer the one technology over the other?
2) was there something about the particular application that brought out the best or worst in each one?
3) when comparing each type of technology, were designs of approximately equal design quality being compared? (i.e., a decent analog delay against a decent digital one)
4) Is there something about the design of the one that could be mimicked by altering the design of the other?
I know my own comments were not prompted by any belief that one is *better* than the other. Rather, they were prompted by the recognition that at this point in time, one could be made to sound pretty much like the other for much less money and effort Will it be possible to detect nuance differences? I suppose, but no moreso than two different builds of the same pedal using unselected components with nominal values.
Of course, if MN3005's could be had for a buck apiece, and PT2399's cost $10.95, I might be singing a different tune.
Quote from: Transmogrifox
So even though many commercially made digital pedals fall short of the mark, it IS possible to do it digitally and inexpensively.
Guys who design stuff always say that, but I'm just saying I have yet to hear it in A/B tests with *The Real Thing*. I'll be happy when I do. A cheap Roland Space Echo in a stompbox woud be nice :wink:
Kerry M
:shock: On another forum some hardcore tape-echo fanatic said "The best thing about real tape echo is that it is so bad", I think he meant that certain "lo-fi" and cascaded natural sounding filtering, distortion and flutter when echo is fed back makes them work well and sound natural
But I believe in this analog versus digital discussion is no end and no right and wrong answers excists, if this forum is up next year someone might argue the same thing again and again. If done properly, digital can sound good and performing musicians can use them with success.
Mark,
I'll address your questions later tonight when I'll have a little more time.
But for now think about this-
What endears me to analog delay is the character and un-predictability.
Said another way- Analog delay is a musical instrument itself,
it reacts to the player, especially a "full featured design".
Any time digital is designed for "character/un-predictability", it comes off sounding fake, irritating or even said to have a glitch or bug by players.
More later, Thanks, Ed R.
Quote from: NasseBut I believe in this analog versus digital discussion is no end and no right and wrong answers excists
I don't mean to imply Analog=good, Digital=Bad, or that digital delays aren't musically useful. It's just that it is often claimed that the *charm* of an analog delay can be reproduced easily enough with cheap digital technology. I find these claims are often made by people without ready access to analog delays. When somebody can A/B a digital delay with a vintage analog delay and say they've got it, then I want to hear it. Obviously digital is *close enough* for a lot of situations, and plenty of people plain don't like the sound of analog delays, but if it's possible to make a DDL as chewy and gnarly as my analog delay, let's hear it.
Kerry M
Pete Snow and I had a delightful lunch today, and now that he's safely returned my POLYPHONY collection, I'm going to have to scan and post the 1983 Craig O'Donnell article on "Why Spring Reverb Will Never Die".
Yes, analog IS wonderful for the unpredictability and nonlinearity it can add, and sometimes what you want out of analog IS its lack of "aural hygiene". I see that advantage of analog over digital in so many domains of instrument processing, but when it comes to delays, any sonic advantage starts to evaporate. Even *if* analog nudges out digital tonally, the capacity to have things repeat many more times, and run delays out past 400msec, turns it into a qualitatively different device. I put together one of Dean Hazelwanter's HT8955 boards and having 4/5 of a second allows you to think differently about where the delay fits in your music. It stops being an embellishment and starts becoming a partner. I'm not saying I would tolerate ANY tonal degradation to achieve that, but in a cost/benefit analysis, whatever it is that one loses by using something like that chip, as opposed to an MN3005, is more than made up for by what it buys you.
Of course, start talking about flanging or distortion, and I become unapologetically and unreservedly analog in my orientation.
Transmogrifox, Nasse, both of you talk about tape delays.. have you ever tried buildign one into a wah case.
you can use the many recorders found in pawnshops.. or steal the guts of the fullsize tape units and cram it in a wah or volume pedal and use the pedal to change the speed of the tape motor.
palce a pot on the side and a switch there or use a push pull so you can select either the rocker to do the volume of the delay or the speed
Quote from: AnsilTransmogrifox, Nasse, both of you talk about tape delays.. have you ever tried buildign one into a wah case.
No, I haven't tried it. Have you? I have been scared away from the task of tape delays because of the mechanical implications. Have you discovered a quick and easy way to mod a tape recorder as such to record and playback simultaneously without having to get overly creative in the mechanical box modding?
Also, what does a tape head need in terms of preamp and impedance considerations?
Any good ideas about how to implement the tape itself?
Then I can start thinking about a wah shell...
Ansil wrote:
QuoteTransmogrifox, Nasse, both of you talk about tape delays.. have you ever tried buildign one into a wah case.
No, I dont! If I would I´ll prefer bigger unit sized like small amp top, like old tape echos, maybe so huge that I can put spring reverb in too. Lots of knobs pushbuttons and maybe translucent case with some cool color lamp or leds inside... Just got copy of old 60´s maq article "Do It Yourself tape echo with tubes". Ansil, drop me a PM if intersted :wink:
When I played in a band 30 years ago, we got somewhere free old mid sixties Meazzi Factotum (Made in Italy) vocal amplification system, all tube 2 x 22 watts EL-84 power amp. Yes, you guessed right, sounded good with guitar. It had built in multi head tape echo. Later I messed with some then new Dynacord tape echoes, and old Vox "Long Dick" tape echo. Since then I have owned only few cheapo BBD delay pedals and some cheapo digital delays. Summa summarum, all delay gear that has been available for me, has been either very old, in bad condition and seen their better days (tube/tape delays) or cheap ones with somewhat low specs (my digital and BBD stuff) aliasing and hiss included.
New better-class digital gear, IMHO are better than those old battered Meazzi and humming Long Tom in bad condition and my cheapo aliasing BBD:s, even though if you watch for specs of high end analog and digital tape echo simulators you see Meazzi and Vox Long Tom "programs" mentioned.
Maybe good analog one is better, but not cheaper and does not have as much presets and midi control.
well,i think that trying to compare a POD,or another DSP based delay to an analog one,is a different business than what we were discussing here...Mark is trying to stress that ,if you BUILD (not program or model in an algorithm) the same filters,the same pre and de emphasis,the companding,etc,then sound quality between a bbd and an 8 or 10 bit delay line,is difficult to tell...i did some ear testings today between my old memory man,my pt80 and one 8 bit delay,that has the same supporting circuitry than a boss analog delay,and i can´t find the difference...but maybe my ears are not as educated as yours...
Well my input is, I have an Ibanez PDM-1, BOSS RV-3, Korg SDD3300 (what a monster, 2 full rack spaces!), and a 1967 Maestro Echoplex. I love all of them! The SDD3300 has a preamp section to die for and it does things that only God could make happen before it. The PDM-1 must be pretty good, it is Carlos Santanas only delay unit he uses and has been using since the late 80s it has 5 JRC4558 IC chips in it, yes that is 5 of them babies! It sounds heavenly. The RV-3 just sounds damn cool at the right setting. Can sound just like a slow gear at the right setting. The Echoplex is well, pure diamonds and gold. Hell I love all of them! They all have their own cool sounds and personality.
Has anyone figured out just what is it in Maestro effects that pass on that strange heavenly tone that all Maestro effects pass on to any signal going through them? I seen this question brought up on many forums in the past. Many folks report the same experience with Maestro effects. A strange tone sweetening effect not found on any of their other old vintage effects.
Well, the simplest way I can think to educate your ears is to do the following.
You'll need a sinewave signal generator, headphone amp, and a willingness to disconnect the dry mix resistor from the final mix stage, so that you'll be listening to Only the delay signal hrough your headphones.
Now start with 400hz, set delay time at max, and take a listen- you'll find wet only, to sound pretty close to dry.
bump down to 80hz, first sign of a problem, low freq takes on a buzzyness
bump up to 4khz, now regardless of the LPF cutoff freq. there will still be still be some signal, and it will have "artifacts" not present in the dry signal, and if you sweep your signal generator from say 1khz to 6khz
your ears will be amazed at the "un-natural" junk riding with the signal.
A poorly designed BBD based delay will also have "junk", but different sounding.
I hope this is clear, but I'm no writer or debater, I would expect makers of digital gear to take offence at my comments, but that it Not my intention. digital has it's purpose as a looper, no argument......
analog BBD done right, is best for echo.
Ed R.
Thanks very much for the information,i've learnt a lot from this topic
Mark,
From your earlier questions-
1) What it is I prefer in analog over digital, is analog BBD technology
suffers from only one of the two forms of "un-natural distortion" that digital suffers from.
Both suffer from "aliasing or foldover" distortion, which is created any time the input frequency is 'higher" than 1/2 the sampling frequency.
This creates the "tinkling, ring-mod" undertone most noted riding on the decay.
In both technologies the cure is- never let the sample rate enter the "audio range" times 2. (or 3)
this is easy now with modern digital, hard with BBD's, but possible.
Digital suffers alone with "quantizing" distortion, which caused by the fact that each sample of audio is assigned a "number value". for example if the number values were 1,2,3,4 etc. but the audio value is actually 2.1 or 3.7, the digital process will round up or down, in this case to 2.0 and 4.0,
this causes 'quantizing distortion" on the playback.
This creates the "gritty" undertone, audible at all frequencies, not easily filtered. And can be heard in any digital process under 8 bit.
2, 3 and 4)
Digital delay Can be done right- best example would be the Korg SDD series. I'd love to get my hands on one- to see for myself.
I've said enough, I meant to offend no-one. my apologies.
Thanks, Ed R.
QuoteI've said enough, I meant to offend no-one. my apologies.
:o Well I think
I poured gasoline to the flames claiming there is some useless debate when people were discussin bout fine art of delay and echo effect :o
But anyhow hope discussion continues under this or other topic...
And original question, chips like those cheap and cheerful PT2955s, can they sound good...
Quote from: Ed Rembold
I've said enough, I meant to offend no-one. my apologies.
Thanks, Ed R.
I don't think anyone took offense, and I don't see where an apology is needed. Everyone has their opinions. You went a step further and gave some good information behind those opinions. This was a totally enlightening and interesting discussion, frankly, I'd like to see more like these.
This thread came along at a good time for me. I just picked up a beat DD-2 at a pawn shop for $40 so I figured I could safely muck with it without drastic consequences if I did something horribly wrong.
On the DD-2 the outside lugs of both the echo volume and repeats/feedback are connected together. Lug 1 (CCW) is ground. Putting a 0.01uF cap across these lugs gives a slight rolloff of the treble on the repeats but it doesn't accumulate (each repeat doesn't have less treble). I tried increasing the cap size, as high as 1 uF, which increased the treble rolloff but also decreased the number of repeats with the feedback control at maximum. Using a 1 uF cap I got maybe three repeats vs dozens with no cap.
Putting the cap between the wiper of the echo volume and ground had no effect and between the wiper of the feedback and ground was the same as between the two outside lugs.
If I could get the repeats to loose increasing amounts of treble without the loss in numbers of repeats I think I would mod it permanently but as it is I don't think it's worth the bother.
Incidently there is just enough room in the DD-2 to mount a mini toggle just above the echo volume control, beside the "effect on" LED. If the psuedo analog mod doesn't work out does anyone (Mark?) have any other suggestions for a worthwhile mod to this pedal?
in a nutshell i just set two of them side by side and fed the tape through both heads.. used onboard preamps
:? Many "better-than-average" digital units seems to leave something desirable features not included, like high enough sampling rate, eq/filtering settings for various echo/delay effects and feedback/multitap facilities...
So an ideal delay machine would be one with high delayed signal quality, and steep and versatile filters with huge adjustable range of corner freq...
Anyone remembering Graig Anderton "Mudguard" project (steep highpass filter to remove "mud" from your reverb/delay, and the euro/brit "instro/surf" and rockabilly players swear by lopass filtering...
Rick, My DD-2 gives me almost infinite repeats with maximum feedback ;) You could try playing with the trimmers inside ;)
I get almost infinite repeats without the cap mod too. I would like to know why adding the cap so drastically cuts down on the number of repeats. My guess is that by cutting out some of the signal with the cap there is less to amplify on each repeat. I think if I tweak the trim pot for more repeats with the cap mod it will get uncontrollable when the cap is switched out, or at least severly reduce the usable amount of pot rotation.
I believe there's a clock trimmer there that will allow you to set the delay time a bit shorter/longer. I tweaked it a bit, but make sure you mark where the trimmer was at first! When I was done tweaking it I got a little longer delays, not much though. I'm too lazy to do any really extensive modding :mrgreen:
ed r.
you post a very tight case. i have 2 roland sde 3000 delays that i wouldn't
trade for anthing. some very hard hitters still use them . i'd say they're about 20 yr's old. i'd also say you know more tech stuff on this subject but check it out. i've had pro's offer me what i paid for them, of course they are in mint condition. $1000. bucks each.
http://www.theorangepeel.net/inside/SDE3000.html
peace,
tom,,,,,,,tom......tom,,,,,tom...to......t............................. :wink:
24W for a delay! WOW! :) That's more than my practice amp :lol:
Rick,
Yup, cutting bandwidth also cuts overall amplitude. Some designs, of course, have recirculation trimmers where you can offset amplitude losses with a trimpot. Ideally, what one wants is a full schematic could indicate where to insert a proper active unity-gain lowpass filter, maybe even a 2-pole job. Admittedly, the cap-to-ground solution is primitive. I've had good luck with it. Not sure why you seem to get an all-or-nothing outcome.
Ed,
Don't be too strongly persuaded by the textbook sampling formulae. The reason one wants sampling of at least double the highest frequency is that it is difficult to accurately reconstruct a waveform without having at least one sample to reflect the positive half-cycle, and another to reflect/sample the negative half-cycle.
Of course, there is no divine law which says that the peak of the positive half-cycle will occur at the precise moment the clock tells your gadget to grab a sample NOW! (and ibid for the negative half-cycle). For that reason, of course, one wants as many samples per wave-form as possible with as much resolution as possible. The resolution in terms of word-size (8-bit, 10-bit, 16-bit, etc.) reduces the likelihood of quantization error as you correctly point out, and the resolution in time (which is essentially what sampling *rate* is) assures that the actual changes in time to the signal are faithfully depicted. All waveform changes should be accurately reflected as far as their "height" is concerned (no serious rounding up or down), and no changes should occur in between samples. The more samples you take, the less the likelihood of any changes occurring between samples - i.e., keep watching and you don't miss anything.
The thing one needs to remember, though, is that the clock rate is fixed, but the frequencies the device is sampling vary all over the place. Whether analog or digital, a 20khz sampling rate does a piss-poor job of "describing" a 10khz waveform sitting atop other content. On the other hand, a 20khz clock rate (and whatever delay interval that corresponds to with the chipset in use) does a pretty decent job of representing a 500hz waveform, with little error (or rather, MUCH much less). Indeed, it is a truism that for any sampling rate waveform fidelity will increase as audio input signal frequency decreases - the slower the waveform changes, the easier it is to keep up with.
This is all the long way of saying that the quantization error and other sources of distortion stemming from the act of sampling at a fixed rate with a fixed word-size, create problems in proportion to the bandwidth one is aiming for. Aim low enough (e.g., 1khz bandwidth with brickwall filters) and the problems all go away, assuming the word-size is acceptable (e.g., I wouldn't expect any such "filter miracles" with 4-bit sampling). This is why imposing steep lowpass filtering to "warm up" a digital delay renders it acceptable and indistinguishable from analog to many users. What that "magical rolloff" might be with vary depending on a bunch of parameters, but I have no doubt that there will always be a filtering configuration that achieves the goal of blurring the lines between A and D.
Analog samples WILL leak if contained too long (i.e., clock rate too slow), so they are subject to their own type of quasi-quantization error. And if you recirculate the signal, that error is multiplied. But of course, you can repair/mask that with filters. Mike Irwin designed an ultra-long analog delay for Modcan, and he tells me that if you set the bandwidth ridiculously low, but use very good filtering, you can have the thing recirculate for a half an hour and "still recognize" the original signal in there. Of course, a human brain recognizing it as somewhat similar to what you started out with is a very different thing from having it look similar on the scope.
Admittedly, there is a tendency to aim budget delays a little higher than they ought to be aimed (i.e., striving for more bandwidth than the filters, sample rate, word-size, and application might call for) and that WILL introduce objectionable qualities to the sound. I guess the suitable analogy is this: take a picture with a webcam and blow it up to front page headline size and it will look absolutely terrible in comparison to a blow-up of even the cheesiest disposable 110 camera. Make them both B&W 1.5" x 1.5" images, though (like the little head and shoulders shot in a column byline), and whatever image quality differences do exist between these technologies will completely disappear.
And Ed, you could only offend me by losing your enthusiasm. :)
explaination of "apology"....
Sometimes I forget that when I post something here, that I'm not talking to a few friends, but rather to the world.
We have vetran dsp programers, expert digital designers, owners of excellent rack gear. etc.
And my focus at times becomes too narrow, I tend to just focus on the little world of analog pedals.
Great comments/points were made by all.
I have heard/tested all the common available digital delay pedals.
but not all the best rack digital delays.
I do still own a Marshall JFX1 (digital rack delay) I used to think it was great, until I was able to compare it to an analog delay of equal delay time- then and only then, could I tell what I was "missing", and then and only then, did it's "digitalness" become So apparent.
Ed R.