Based on memory the PCM60 didn't have the spin and wander capabilities that the later PCMs had. I think the later PCMs, possibly starting with the 70, used a LexiChip to create the tails. Those were custom designed DSP chips Lexicon made. The PCM80 had a LexiChip and a Motorola 56002. The PCM90 had a pair of LexiChips.
Perhaps I’m doing something wrong but my test of my hardware PCM60 (as described at 2:38) does NOT null on the test. My hardware has the Lexicon eprom update from the factory. Maybe they added something in that hardware update? EDIT: to clarify, I tried it again with several of the algo's in the PCM60, and none of them nulled. It wasn't even close actually. So yes, there is definitely modulation happening in the hardware PCM60. Whether that was introduced in the Version 2 firmware update, I can't say though.
@@slamcrank I’m sending a PT click track out to the PCM60. Recorded a stereo return back into PT. Repeated tor the second takes. Inserted a 1 band EQ3 on both. Flipped polarity on one of the playbacks and it all disappeared completely.
@@zagatoalfa Weird - I'm basically doing the same thing... click track out to PCM60, same for the repeated take. I use the "trim" plugin-in PT to flip the polarity. There's quite a bit of modulation left over that does not null. Do you know what firmware version your unit is on? Should be noted on the back, or if it was updated later, there may be a handwritten note on a sticker on the back or bottom of the unit.
The PCM70 is very similar to the PCM60, with the same ARU chips, but the PCM70 has an additional dedicated Z80 processor that can constantly write into the DSP control store, and thus create the modulation of the reverb. Both the PCM80 and PCM90 use Lexichips instead of the earlier CMU/MMU/ARU , but also still appear to be using a Z80 for creating the modulation. The PCM60 is incapable of modulation, it just doesn't have the hardware for it.
I embrace the philosophy of "if it sounds good it is good" and happily accept that in the mix things often sound different than expected and subtleties such as modulated reverb tails are often insignificant.
Hmmmm these days I wonder, so much detailed digital editing and adjustments just to get something sounding interesting. I can't be bothered caring anymore lol. Much a do about the same stuff.
I’ve become drawn to your videos as much for your process in critical thought as for your gear and production expertise. I love that everything appears well-pondered and tested-even your own feelings.
There’s an old forum post from Sean Costello (Valhalla DSP): From a theoretical point of view, both Altiverb and the PCM60 are LTI (linear time invariant). So, you could get an impulse response of the PCM60, and it should - theoretically - sound the same as the PCM60. The PCM60 only has a few possible settings, like 32 or 64, so a set of impulse responses should cover all the functionality of the unit. Many of the other Lexicons are not linear time invariant. The classic Lexicons, like the 224, 224XL, 480L, and the PCM70 Concert Hall algorithm, have lots of time variation in their algorithms, in the form of moving delay lines or time-varying output taps. The resulting sound is very rich, and can't be captured by convolution. Many of the Lexicon reverbs also have a form of "program dependency," where different input volume levels result in changes in the reverb time and other parameters. This would probably be described as "nonlinear," and also isn't captured by convolution.
@@Otocol yes, anything non-linear (distortion/saturation) or ever-changing (modulation, dynamics or any kind of randomness) cannot be captured in a standard IR. that's also why I use Celestion's Speakermix Pro to simulate guitar speakers dynamically rather than the static IRs that almost everyone else uses. I've heard the difference since cab IRs became popular, they always sounded and felt 'dead' to me.
@@desperateBeauty Yes, the completely static cabinet IR have very distinct sound from a dynamic counterpart. Robotic, in-organic or, indeed, dead are good descriptors for it. However, that's what I use myself. To mitigate those issues, I use two or three different impulses that I drive differently based on the level of incoming signal. Then I add some very slight randomization in there too after the reverb to make it more alive. Sometimes even before the reverb to add some small variability on how the signal interacts with the IR. And when using midi, then I can easily randomize even the reverb's settings slightly for each keystroke, but keep the values static during the keystroke to prevent artefacts. With FL-Studio this can be set up with Patcher for easier subsequent use, or to use some complete ecosystem like Kilohearts plugins. I'm not sure if other daws has something alike Patcher to make it easy. Just my two cents...
@@anteshellsounds fun. I did play with 'dynamic-izing' IRs in the past in a similar way, and while I got fun results, it never sounded quite like a realistic paper cone which is what I really wanted, at least to start with. but yeah the possibility for creative dynamic sounds is endless.
I love the in-depth analyse and deep understanding these videos convey without any clickbait , the whole atmosphere and professionality... Still the best there is in this class..
What you said about an interface for IRs is why i adore two-notes cabinets. Every cabinet has a pic and a story and is sorted by type, you can adjust position and mic, its just a nice way to use an IR i could probably get free, but wouldnt use.
The biggest enemy of good impulse responses is the convenience of using them. So much better to see what you're getting than to remember a long list of filenames or listening to them individually.
Dan, if you want to check whether a reverb modulates or not, you can do that very easily. Just feed a _static_ oscillator (f.e. a sawtooth - absolutely stable in pitch) into the reverb and hold it for about 3 seconds or longer (for the delay pattern to settle). If the reverb doesn't modulate, you'll end up with essentially a combfiltered signal where your can't perceive any room impression at all - not while the note is held, nor in the tail when the note is off and you only listen to the pure tail (you can also set the reverb to 100% wet, which clearly let's you hear that). If _that_ is the case, you can perfectly capture the reverb with an IR of it. If it _does_ modulate, you would hear a constantly appearing seperation of the reverb from the original signal (much like some form of a chorus - because that's essentially what it is really) _during_ you feed the oscillator into it and quite possibly in the tail afterwards too (If the modulation is indeed modulating the delays in the feedback path). Then you _won't_ be able to capture the sound of that reverb unit no matter what - except if you are able to turn off the modulation. However, mostly then the magic often is gone then. If you're using non-modulated reverbs on short, percussive material, that issue is reduced and might not matter too much. But when used on tonal material that lasts longer - especially without a lot of change in pitch (like synths in contemporary music often do), modulation (if correctly implemented) can do the magic trick and you very often simply can't get away without. Whether the PCM 60 does modulate in some way or not i don't know (never had one and also never had the chance to work with one) - but i doubt it. Afaik, the first Lexicon reverb that featured modulation was the PCM 70, followed by the L224 (someone by all means correct me if i'm wrong here) Hope that helps.
Hi,im the developer of uberverb ,(midiverb emulation),when i developed my plugin i had a choise use convolution or true algos...i choosed algos,took me a long time and sure i could use IRs and make my life a lot easier but to me IRs are static,lifeless like a photo compared to a movie....and if i wanted to implement true modulation later? Or inovate my algos? thats why i would never choose IRs on the first place.
@@DanWorrall No worries! Sometimes our mind just locks onto a certain aspect and we completely overlook something else. Terrible when this happens when you're troubleshooting something and you looked at every tiny detail for hours, just not that one thing your brain had already ruled out. 😋
i wasn't impressed with this plugin, but I'm glad plugin alliance is making more reverbs. as far as IR stuff, yeah i feel that! curious how many plugins these days are just IRs. The blind test i heard the subtlest of difference, but ONLY on the drums and only in the longer part of the tail. it also took a bunch of listens. Those differences don't matter and neither were better than the other, just different (nor could i tell you which was which). I like how you talk about the value of workflow in the bettermaker and that not all IRs are are equal. thanks for the video Dan!
It's so delightful to see Dan constantly releasing new videos, it shows that his health is improving as well. In a scenario where I assume Dan has a kidney failure, if he continues to produce video content, I would willingly give him one of my kidneys without hesitation.
It could be worth checking out the Wave Alchemy Pulse plugin! I'm pretty sure it uses IRs as well, but it has more tweaking options (including a gate) and is a third the price.
The simple point and purpose of random modulation in the tail of a digital reverb is to prevent a ringing that occurs when the same pattern of delays repeat over and over. They are not the only reberbs to use it but whas first in modulating left and right separately with a third or more instances towards the center of the stereo image with most of the low end rolled off to make the center sound further away thus creating that nice depth Lexicon are renowned for.
I heard the modulation difference. I'd be unhappy to be stuck with static IRs if the hardware did have modulation. it doesn't have to be a massive difference, it's a difference in feel which is often what we're looking for.
I'll need to check this a bit more, but to me, for the A/B comparison, it sounds like B separates itself from the source more than A. It's subtle, but if you're trying to create front-to-back depth, then it adds to the effect.
Pcm 60 and pcm 70 were my IRs I used and I loved them. The 224 and 480 to me relies so much on saturation and modulation that IRs just don't sound that similar to me. More modern reverbs rely less on modulation to hide the issues
Also it's worth pointing out that BM60 does indeed model some weird input ad/da clipping and also strangely if you crank the output level knob to full you get a lot of grabby punchy not at all subtle compression happening. It's a strange plugin indeed.
People just like to sh!t on Acustica. Personally I have a few of their plugins and they’re useful. Used to work in a place with a real 670, and the Acustica is the only one that gets the sound quality right (The UA one is close). You’ll notice the vast majority of people whining about Acustica haven’t used their stuff. (Nor do most of them know what a real 670 sounds like)
Obviously Acustica stuff is a niche. It’s convolutions of real gear. Which you don’t always want. But when you do, it’s quite helpful to have in your toolbox. Acustica is dynamically morphing between many IRs of a given unit, depending on the source material. It occupies a place between a static IR and an algorithm. For me, having a representation of actual vintage Neve EQ, or a rare desk made for Polygram, is helpful. I would estimate I reach for an Acustica thing about 10% of the time. More often choosing UA or Softube. But when I want the Acustica thing, i really want the Acustica thing.
Just a thought - an algorithmic reverb with modulation still might result in a perfect null if said modulation is deterministic It might be a good idea to run the null test on rendered files made from items from two different locations on project timeline (to rule out cases where modulation-determinism is project-time dependent)
I thought of that and tested renders as well. I didn't try different time codes, but honestly after I tested the stereo behaviour I had no doubts left that it's IRs.
@@DanWorrall Oh yeah.. I thought so too (I suggested so in a comment on the previous video) I actually meant it regarding the tests you made on the lexicon plugin algorithms (to say that those that presented a perfect null still might have a modulation component to them) I’m guessing probably not because it will be kind of funny to have some modulations deterministic and others free running on the same plugin.. then again not impossible (say if one of the algorithms requires a very slow modulation and the other a fast one - this might be a good enough reason for a developer to include both options for different algorithms on the same plugin)
The Random Hall is definitely nicer than the IR. I was pretty confident during the blind test. A was certainly usable too, B was nicer, thicker, more.... well, moving.
Yeah I thought the same. To me A sounded more like it had really loud dither in the tail, where as B had a sort of "wobble" to it, which kinda made the white noise sound in the tails a lot less offensive, and sounded a lot like oldskool organ physical modulation, where a card/plate is span over the end of the pipe the sound is coming out of, kind of like a 360 deg throttle body for the sound pipe (if that makes sence), it really suited the instrument choice in this vid too, which I think helped a lot.
My favorite reverb plugin, IK’s Sunset Sound, is convolution based. There is apparently analog modeling for the signal chain used in Sunset Sound but the room sound itself is all IR. If you like the sound of a reverb and it gets you results fast, who cares. I agree on the BetterMaker; the interface, the collection, quality and layout are worth it if you like that sound. Reverb is one of those things I always start from a preset, then tweak to taste, so user experience is everything to do that quickly. I also find I only really prefer modulated tails when I’m really going for a noticeable “wash” effect.
I tried really hard to dislike this plugin (as I have so many reverbs and didn't want to buy it!!). I tested it against all my UAD reverbs & QuantX by Relab (and various others). In the end I gave in and got it - WHAT A REVERB, superb. Thanks for your testing & thoughts Dan
Interesting, in your Lexicon plugin suite vs Impulse A/B blind test, I actually heard a definite difference but much preferred the impulse! Not sure why, but on my headphones here at home something in the stereo field or some low-mid frequency felt a bit annoying in the B example. Very interesting! This makes me wonder if some reverb plugins and settings that annoy me can actually be improved by simply capturing a static impulse.
This is why i just use an IR of classic units. Past 2 Future Reverbs has a lot of great IRs of classic units like Bricasti, Lexicon etc. A good IR reverb has an option to add some sort of modulation to the loaded IR. Convology XT (free) and Fog Convolver (not free but excellent) can do this.
I tested this one when it first appeared in my email feed, and decided to pass. There are too many options out there and if I grabbed a new reverb it would likely be ReLab Quantex - it sounds very good to my ears and loads presets more quickly than my current favorite which is the ReLab 480. For me, IRs are too static.
All I know is that I was Greg, I would've probably shit my pants. Like of course it would be cool as hell to know that Dan found something that I made, use it for a video and mention me by name so I could get some recognition for this work I put into it and shared with everyone.. Yeah, that's cool, but what would make shit my pants is for Dan "THE MAN" Worrall to say to little old me "Way to go" in front of all my friends. Lol. WAY TO GO GREG!!!
Dan, do you have Meldaproduction's MXXX plugin? I fell in love with Alesis' Quadraverb (quality check seal from 1989) and if you want I could make a plugin just like this (GUI of the real hardware), including modulation options, within the MXXX frame. I wanted to do it for myself, anyway. The only catch is you'd have to have MXXX. (which I only have because I subscribed [to own!] Melda's excellent plugins)
The difference between modulated and non-modulated reverbs was rather noticeable on drums to me, cause I could hear how reverb tails sound in isolation. And what's weird is that the modulated one's decay sounded smoother?
Hi Dan! In your channel membership, what's the difference between lowest and second lowest tier? And can you make a tutorial on how to create impulses inside the Daw? Like not a reverb but an impulse I already have+EQ so I can import it directly in let's say BiasFx2?
Afair no difference. People requested an in between tier, but I couldn't think of a way to differentiate it. If you want to EQ an existing impulse, just EQ it: bounce it through the setting you want, then load the result into your convolution plugin. As long as it's linear it should work.
Strange how it feels a bit fake when it turns out to be impulse responses, even if they sound perfectly fine. I have some very nice reverbs, but in practice I always grab a simple one first, otherwise I'll be twisting knobs for the following half-hour looking for perfection. A set of quality go-to impulse responses might just be a great thing to have.
Hola Dan, tus videos son muy utiles y educativos, los subtitulos me ayudan a entenderte mejor, podrias hablar de los plugins de Airwindows? Creo que podria ser interesante para todos. Saludos y gracias por tantos videos 🙏🏻
I wonder if they put some kind of bit-reduction on the input of that thing....or sample rate limiting. Like 8bit or 20k sample rate. Point being, the original unit is probably pretty limited in that regard.
Can't remember exactly, as I've briefly played with a real PCM60 maybe 25 year ago and I haven't done any dissection of its internal algos. Even if the modulation will be there (I won't bet on it), it will be very subtle, not anything remotely similar to higher end units, where you can get its famous modulated tails. Also back then in guitar rigs those more affordable Lexicons were often combined with modulated delays, to make it like rich sounding (and more expensive) PCM81 and 91 units.
Hi Dan, i have a very interesting topic so like it would make for a great video if such thing exists Here's the setup I'm aiming for: - EQ Instance One: Placed before the input of Amp 1. - EQ Instance Two: Sidechains the sound of Amp 2 to EQ Instance One and EQ Instance Three. - EQ Instance Three: Placed after Amp 1 and sends feedback to EQ Instance One to match the sound of Amp 1 to Amp 2. I need an EQ VST that can handle the following process: EQ Instance One should iteratively adjust the EQ settings to account for the distortion and unique character of Amp 1, ensuring that the output sound of Amp 1 matches the target sound of Amp 2 as closely as possible. Disclaimer: I'm not looking for an amp sim with tone match. Do you know anything that fits these requirements? Or maybe another way of doing such a thing without spending weeks trying to recreate a tone? Im really stuck and frustrated :( haha
So, when it comes to this being worth it, I have to take into account it’ll be $30 at some point in the future. I’m currently happy with my reverb options, but I can see the value.
Another great video, thanks, just a small aside, given the impossibility of generating random numbers with a computer, and thus random noise within a reverb tail. To a programmer, it is entirely possible to use a random number generator, without reseeding, and getting an identical effect every run. It is by all accounts still random. Beware though, in the cave that is random number generation, there be Dragons!
@@DanWorrall But the same code could be run, one version seeding every time and the other always using the same seed, both would sound the same to our mortal ears, but the one that is seeded would use up the computers entropy pool, if you want it to really be random that is, or as close to as is possible. Most sane programmers would I think, in this case, just use the same seed each time. The version that used the same seed would null.
@@DanWorrall The most basic example is the function rand() in the C standard library, if you dont seed it with the time, it will produce the exact same set of numbers every time.
You could test for that by bouncing the test files at different timecodes. But I'm pretty satisfied it's just IRs now, and there's not even any pseudo randomness.
I am interested in getting the BM60, but I will wait for a good sale. :) And Dan, I don't know why you are not making big bucks doing voice/narration work. Or are you? :)
Great video as usual. I’ve wondered whether the A/D stage would have any impact on the all pass filter network to create the reverb and whether IR’s are suitable for this, or if there is some kind of trade off. Presumably the IR is capturing some linear aspects of this stage, but it’s not possible to sample the reverb without it. I’m also not sure whether the plugins own input saturation is going to work as the hardware would. I find IR’s to end up sounding quite stuffy in a mix, particularly if a lot of sources are feeding it. For individual tracks or certain uses, it’s not an issue, but some modulation or pitch change can add some excitement. Valhalla Vintage and the various 224 plugins should share some similarities with the PCM60 algorithms, although they’d be harder to dial to the specific PCM60 settings. IIRC Avid Revibe II has some PCM60 inspired presets too.
IRs would include the filters in the AD/DA converters, but not any non linearities. The input distortion in the BM60 must be separate code as IRs are linear.
I do NOT think the PCM60 has "the" Lexicon sound. So it would not surprise me at all if they don't have that tail modulation. Edit: And yeah, both reverb A and B sounds to me like a bunch of densely packed echoes, just like the PCM60 does. And the MPX, and LXP etc. While later PCMs, and the 224 and 480 etc, there the tails are just one smooth continuous sound.
I'm wondering if there might still be modulation happening and that different instances of the same plugin might simply be pulling from the same seed at the same time, so they'd effectively have the same randomness applied to them. That way they should still null perfectly. This is just an idea though, and I could be entirely wrong with this.
No. What parameters would you modulate? Closest you could get would be some kind of chorus or phaser type processing for the wet only signal, but it's not really the same.
wonder how you feel about arturia’s lexicon emulation - it’s an emulation of the 224 and not the pcm 60, however. I never used any of the original hardware but it definitely has a flavor I don’t get from the valhalla reverbs (which also get a lot of use from me) I might pick up the bettermaker once it goes on sale for lower than the current loyalty pricing of 70 bucks. if it hits 30 I could see it being a worthy impulse buy, no pun intended.
Look I love arturia stuff. But that reverb did not win me over. I have the UAD 224 and it IS the 224. A direct port of the algorithm and modeled i/o stages. The arturia is new and has some mod cons, but I couldn't get it to sound as good. That could be me, but I think why bother with an emulation when the realm thing is already available?
If you want to get Valhalla to sound closer to Lexicon 224, pick Dirty Hall or Dirty Plate modes with 1970 color and put bass multiplier at x1.00, also relax some high frequency damping.
I believe the famous Lexichip (with reverb modulation) came in the units after, and the last Lexichip unit was the MPX-1. However, the Lexichip was the same between units, but the PCB routing and extra chips differed. Such as the MPX-1 has a full multi effects chip as well. The Bettermaker is a 1Gb download. False marketing imo. I'll probably pass even when it is $19 or $29 in the future. Have enough reverbs already...
Does it matter… well - back in the day sources were summed going into the reverb so it’s not like there’s the possibility that multiple reverbs at the same time have modulation interferences wasn’t really a thing either. So… nope
If the original unit nulls it's self, then it's IR's should null too, even if multiple were taken. So it's highly unlikely they used multiple IRs in this scenario (as it would be pointless), and would only use them in a scenario where the original unit didn't null with it's self anyway. So the fact of it not nulling because of round robin, is telling you the original unit did not null either, and is exactly the behaviour you're trying to tell apart from, by doing the null test.
Just a note: When switching between two audio files/sources in an A/B comparison, human sensitivity to discerning a difference will be better if you flip back and forth in the middle of static audio information and not at the beginnings of measures or phrases (as most people seem to want to do). Since there tend to be transients at the downbeats and at the beginnings of measures, (such as voice and instrument onsets and offsets), those sounds can hide the subtle acoustic differences that the human brain might be able to otherwise distinguish if the switching is done instead in the middle of measures, notes, and phrases where the audio information is being held more constant without those transients. The acoustic psychophysics of this phenomenon are well characterized in the research literature. Cheers.
@@DanWorrall But often the baseline question is whether they can be perceived at all. A real mix may have times when they can be discerned and times when they can't be. Ruling out differences during times that are more difficult to discern leaves the possibility that discernable differences still exist even in a real mix. Great music and videos. Sending appreciation from San Diego. ☀️👍🏻
it would as long as there's no additional randomness going on, then it likely wouldn't (unless the pseudo-random 'seed', ie. starting point, is also the same and triggered at exactly the same time).
The coveted 90s reverb tone would be better - from the PCM 80 (or 81) & the PCM 90 (or 91). Why didn't these plugin makers do a 1:1 code of it. I hear Lexicon & Izotope merged these days.
Don't be scared, that's a very valid point. I did test renders as well just in case. But I'm hearing that the PCM60 didn't use modulation anyway, so I'm pretty sure that wasn't happening here.
One point that I'd like to add is that I personally agree with White Sea's rants about analog-simulated UIs. I hate having a fake rack unit on my screen with useless buttons disposal. I love Valhalla UI, althought I also don't like their crazy planet names of the algorithms. Having said that, the original PCM analog UI is very fast and well done. Maybe a simpler interface, with the same buttons and names would do the job with less screen real state.
yeh would be nice if they just admitted they are integrating ir technology its not even the worst selling point I would buy a half normal plugin and half ir plugin
![[Live สด] การออกรางวัลสลากกินแบ่งรัฐบาล งวดวันที่ 16 ธันวาคม 2567](http://i.ytimg.com/vi/dNTiBBthKgQ/mqdefault.jpg)
Based on memory the PCM60 didn't have the spin and wander capabilities that the later PCMs had. I think the later PCMs, possibly starting with the 70, used a LexiChip to create the tails. Those were custom designed DSP chips Lexicon made. The PCM80 had a LexiChip and a Motorola 56002. The PCM90 had a pair of LexiChips.
Tested my hardware PCM60. It nulls fully and completely.
Perhaps I’m doing something wrong but my test of my hardware PCM60 (as described at 2:38) does NOT null on the test. My hardware has the Lexicon eprom update from the factory. Maybe they added something in that hardware update?
EDIT: to clarify, I tried it again with several of the algo's in the PCM60, and none of them nulled. It wasn't even close actually. So yes, there is definitely modulation happening in the hardware PCM60. Whether that was introduced in the Version 2 firmware update, I can't say though.
@@slamcrank I’m sending a PT click track out to the PCM60. Recorded a stereo return back into PT. Repeated tor the second takes. Inserted a 1 band EQ3 on both. Flipped polarity on one of the playbacks and it all disappeared completely.
@@zagatoalfa Weird - I'm basically doing the same thing... click track out to PCM60, same for the repeated take. I use the "trim" plugin-in PT to flip the polarity. There's quite a bit of modulation left over that does not null.
Do you know what firmware version your unit is on? Should be noted on the back, or if it was updated later, there may be a handwritten note on a sticker on the back or bottom of the unit.
The PCM70 is very similar to the PCM60, with the same ARU chips, but the PCM70 has an additional dedicated Z80 processor that can constantly write into the DSP control store, and thus create the modulation of the reverb. Both the PCM80 and PCM90 use Lexichips instead of the earlier CMU/MMU/ARU , but also still appear to be using a Z80 for creating the modulation. The PCM60 is incapable of modulation, it just doesn't have the hardware for it.
Dan Worrall: "as everyone knows......."
Me: [grabs pen and furiously takes notes]
I embrace the philosophy of "if it sounds good it is good" and happily accept that in the mix things often sound different than expected and subtleties such as modulated reverb tails are often insignificant.
exactly
Hmmmm these days I wonder, so much detailed digital editing and adjustments just to get something sounding interesting. I can't be bothered caring anymore lol. Much a do about the same stuff.
I’ve become drawn to your videos as much for your process in critical thought as for your gear and production expertise. I love that everything appears well-pondered and tested-even your own feelings.
There’s an old forum post from Sean Costello (Valhalla DSP):
From a theoretical point of view, both Altiverb and the PCM60 are LTI (linear time invariant). So, you could get an impulse response of the PCM60, and it should - theoretically - sound the same as the PCM60. The PCM60 only has a few possible settings, like 32 or 64, so a set of impulse responses should cover all the functionality of the unit.
Many of the other Lexicons are not linear time invariant. The classic Lexicons, like the 224, 224XL, 480L, and the PCM70 Concert Hall algorithm, have lots of time variation in their algorithms, in the form of moving delay lines or time-varying output taps. The resulting sound is very rich, and can't be captured by convolution. Many of the Lexicon reverbs also have a form of "program dependency," where different input volume levels result in changes in the reverb time and other parameters. This would probably be described as "nonlinear," and also isn't captured by convolution.
Thx for the follow-up, Dan. I am just here to say hello. So, "hello". Cheers from Germany
Making IRs of favorite reverb settings is a highly recommended trick.
It doesnt capture modulation as far as i can tell
@@Otocol yes, anything non-linear (distortion/saturation) or ever-changing (modulation, dynamics or any kind of randomness) cannot be captured in a standard IR. that's also why I use Celestion's Speakermix Pro to simulate guitar speakers dynamically rather than the static IRs that almost everyone else uses. I've heard the difference since cab IRs became popular, they always sounded and felt 'dead' to me.
@@desperateBeauty Yes, the completely static cabinet IR have very distinct sound from a dynamic counterpart. Robotic, in-organic or, indeed, dead are good descriptors for it.
However, that's what I use myself. To mitigate those issues, I use two or three different impulses that I drive differently based on the level of incoming signal. Then I add some very slight randomization in there too after the reverb to make it more alive. Sometimes even before the reverb to add some small variability on how the signal interacts with the IR. And when using midi, then I can easily randomize even the reverb's settings slightly for each keystroke, but keep the values static during the keystroke to prevent artefacts.
With FL-Studio this can be set up with Patcher for easier subsequent use, or to use some complete ecosystem like Kilohearts plugins. I'm not sure if other daws has something alike Patcher to make it easy.
Just my two cents...
@@anteshellsounds fun. I did play with 'dynamic-izing' IRs in the past in a similar way, and while I got fun results, it never sounded quite like a realistic paper cone which is what I really wanted, at least to start with. but yeah the possibility for creative dynamic sounds is endless.
Bogren sells something similar: IRDx Core. I’m gonna checkout the Celestion product. The Bogren is pretty subtle but I can hear a difference.
I love the in-depth analyse and deep understanding these videos convey without any clickbait , the whole atmosphere and professionality...
Still the best there is in this class..
Came back in 'owned it' and provided yet another strong video Dan.
This is another reason why I'm here for sure.
Thank you once again from 'Straya.
What you said about an interface for IRs is why i adore two-notes cabinets. Every cabinet has a pic and a story and is sorted by type, you can adjust position and mic, its just a nice way to use an IR i could probably get free, but wouldnt use.
The biggest enemy of good impulse responses is the convenience of using them. So much better to see what you're getting than to remember a long list of filenames or listening to them individually.
Dan, if you want to check whether a reverb modulates or not, you can do that very easily. Just feed a _static_ oscillator (f.e. a sawtooth - absolutely stable in pitch) into the reverb and hold it for about 3 seconds or longer (for the delay pattern to settle). If the reverb doesn't modulate, you'll end up with essentially a combfiltered signal where your can't perceive any room impression at all - not while the note is held, nor in the tail when the note is off and you only listen to the pure tail (you can also set the reverb to 100% wet, which clearly let's you hear that). If _that_ is the case, you can perfectly capture the reverb with an IR of it. If it _does_ modulate, you would hear a constantly appearing seperation of the reverb from the original signal (much like some form of a chorus - because that's essentially what it is really) _during_ you feed the oscillator into it and quite possibly in the tail afterwards too (If the modulation is indeed modulating the delays in the feedback path). Then you _won't_ be able to capture the sound of that reverb unit no matter what - except if you are able to turn off the modulation. However, mostly then the magic often is gone then. If you're using non-modulated reverbs on short, percussive material, that issue is reduced and might not matter too much. But when used on tonal material that lasts longer - especially without a lot of change in pitch (like synths in contemporary music often do), modulation (if correctly implemented) can do the magic trick and you very often simply can't get away without.
Whether the PCM 60 does modulate in some way or not i don't know (never had one and also never had the chance to work with one) - but i doubt it. Afaik, the first Lexicon reverb that featured modulation was the PCM 70, followed by the L224 (someone by all means correct me if i'm wrong here)
Hope that helps.
Hi,im the developer of uberverb ,(midiverb emulation),when i developed my plugin i had a choise use convolution or true algos...i choosed algos,took me a long time and sure i could use IRs and make my life a lot easier but to me IRs are static,lifeless like a photo compared to a movie....and if i wanted to implement true modulation later? Or inovate my algos? thats why i would never choose IRs on the first place.
That brief pause while switching is a dead give away.
It should probably have aroused my suspicions earlier, yes.
@@DanWorrall No worries! Sometimes our mind just locks onto a certain aspect and we completely overlook something else. Terrible when this happens when you're troubleshooting something and you looked at every tiny detail for hours, just not that one thing your brain had already ruled out. 😋
i wasn't impressed with this plugin, but I'm glad plugin alliance is making more reverbs. as far as IR stuff, yeah i feel that! curious how many plugins these days are just IRs.
The blind test i heard the subtlest of difference, but ONLY on the drums and only in the longer part of the tail. it also took a bunch of listens. Those differences don't matter and neither were better than the other, just different (nor could i tell you which was which).
I like how you talk about the value of workflow in the bettermaker and that not all IRs are are equal. thanks for the video Dan!
It's so delightful to see Dan constantly releasing new videos, it shows that his health is improving as well. In a scenario where I assume Dan has a kidney failure, if he continues to produce video content, I would willingly give him one of my kidneys without hesitation.
It could be worth checking out the Wave Alchemy Pulse plugin! I'm pretty sure it uses IRs as well, but it has more tweaking options (including a gate) and is a third the price.
Been playing guitar for 30 years and now I’m semi obsessed with Rhodes pianos. I love that simple line you got going on there
Arturia Stage 73, it sounds great right?
@@DanWorrall that's the exact one I was using this morning but I just bought the "Tines Duo" by Native Instruments on sale 20 minutes ago :P
Dan, it feels like Christmas. So many videos from you. I'm very happy!
As long as it isn’t too expensive, an IR plugin like this is quite useful imo
The simple point and purpose of random modulation in the tail of a digital reverb is to prevent a ringing that occurs when the same pattern of delays repeat over and over. They are not the only reberbs to use it but whas first in modulating left and right separately with a third or more instances towards the center of the stereo image with most of the low end rolled off to make the center sound further away thus creating that nice depth Lexicon are renowned for.
Always and Forever really is a great piece. Kinda obsessed with that chord progression.
Oh and I love your content, always relevant, always honest.
Hi and Welcome to the England. burning.
I heard the modulation difference. I'd be unhappy to be stuck with static IRs if the hardware did have modulation. it doesn't have to be a massive difference, it's a difference in feel which is often what we're looking for.
I'm being told that the PCM60 didn't use modulation.
I'll need to check this a bit more, but to me, for the A/B comparison, it sounds like B separates itself from the source more than A. It's subtle, but if you're trying to create front-to-back depth, then it adds to the effect.
Seems like the most important part of any plugin is 1. sound quality 2. ease of use. If it sounds good it is good!
Pcm 60 and pcm 70 were my IRs I used and I loved them. The 224 and 480 to me relies so much on saturation and modulation that IRs just don't sound that similar to me. More modern reverbs rely less on modulation to hide the issues
A real man owns his mistakes.
Thank you so very much for your candor, Dan.
Also it's worth pointing out that BM60 does indeed model some weird input ad/da clipping and also strangely if you crank the output level knob to full you get a lot of grabby punchy not at all subtle compression happening. It's a strange plugin indeed.
You've been Acustica'd.
As in 'I have no idea how Acustica plugins work'.
People just like to sh!t on Acustica. Personally I have a few of their plugins and they’re useful. Used to work in a place with a real 670, and the Acustica is the only one that gets the sound quality right (The UA one is close). You’ll notice the vast majority of people whining about Acustica haven’t used their stuff. (Nor do most of them know what a real 670 sounds like)
Obviously Acustica stuff is a niche. It’s convolutions of real gear. Which you don’t always want. But when you do, it’s quite helpful to have in your toolbox. Acustica is dynamically morphing between many IRs of a given unit, depending on the source material. It occupies a place between a static IR and an algorithm.
For me, having a representation of actual vintage Neve EQ, or a rare desk made for Polygram, is helpful. I would estimate I reach for an Acustica thing about 10% of the time. More often choosing UA or Softube. But when I want the Acustica thing, i really want the Acustica thing.
Acustica, 28Tb files to hold a 3Kb piece of code 😂
Just a thought - an algorithmic reverb with modulation still might result in a perfect null if said modulation is deterministic
It might be a good idea to run the null test on rendered files made from items from two different locations on project timeline (to rule out cases where modulation-determinism is project-time dependent)
I thought of that and tested renders as well. I didn't try different time codes, but honestly after I tested the stereo behaviour I had no doubts left that it's IRs.
@@DanWorrall
Oh yeah.. I thought so too (I suggested so in a comment on the previous video)
I actually meant it regarding the tests you made on the lexicon plugin algorithms (to say that those that presented a perfect null still might have a modulation component to them)
I’m guessing probably not because it will be kind of funny to have some modulations deterministic and others free running on the same plugin.. then again not impossible (say if one of the algorithms requires a very slow modulation and the other a fast one - this might be a good enough reason for a developer to include both options for different algorithms on the same plugin)
Well done. Nice follow-up.
The Random Hall is definitely nicer than the IR. I was pretty confident during the blind test. A was certainly usable too, B was nicer, thicker, more.... well, moving.
Yeah I thought the same. To me A sounded more like it had really loud dither in the tail, where as B had a sort of "wobble" to it, which kinda made the white noise sound in the tails a lot less offensive, and sounded a lot like oldskool organ physical modulation, where a card/plate is span over the end of the pipe the sound is coming out of, kind of like a 360 deg throttle body for the sound pipe (if that makes sence), it really suited the instrument choice in this vid too, which I think helped a lot.
My favorite reverb plugin, IK’s Sunset Sound, is convolution based. There is apparently analog modeling for the signal chain used in Sunset Sound but the room sound itself is all IR. If you like the sound of a reverb and it gets you results fast, who cares. I agree on the BetterMaker; the interface, the collection, quality and layout are worth it if you like that sound. Reverb is one of those things I always start from a preset, then tweak to taste, so user experience is everything to do that quickly. I also find I only really prefer modulated tails when I’m really going for a noticeable “wash” effect.
I tried really hard to dislike this plugin (as I have so many reverbs and didn't want to buy it!!). I tested it against all my UAD reverbs & QuantX by Relab (and various others). In the end I gave in and got it - WHAT A REVERB, superb. Thanks for your testing & thoughts Dan
Why start at the 24th sample when you drew the full-scale sample? (I'm planning to attempt this IR capture process.)
7:13 I legitimately thought he was about to play Yoshikage Kira's theme for the blind test.
Dan is secretly a Jojo fan.
Interesting, in your Lexicon plugin suite vs Impulse A/B blind test, I actually heard a definite difference but much preferred the impulse! Not sure why, but on my headphones here at home something in the stereo field or some low-mid frequency felt a bit annoying in the B example. Very interesting! This makes me wonder if some reverb plugins and settings that annoy me can actually be improved by simply capturing a static impulse.
This is why i just use an IR of classic units. Past 2 Future Reverbs has a lot of great IRs of classic units like Bricasti, Lexicon etc. A good IR reverb has an option to add some sort of modulation to the loaded IR. Convology XT (free) and Fog Convolver (not free but excellent) can do this.
I tested this one when it first appeared in my email feed, and decided to pass. There are too many options out there and if I grabbed a new reverb it would likely be ReLab Quantex - it sounds very good to my ears and loads presets more quickly than my current favorite which is the ReLab 480.
For me, IRs are too static.
All I know is that I was Greg, I would've probably shit my pants. Like of course it would be cool as hell to know that Dan found something that I made, use it for a video and mention me by name so I could get some recognition for this work I put into it and shared with everyone.. Yeah, that's cool, but what would make shit my pants is for Dan "THE MAN" Worrall to say to little old me "Way to go" in front of all my friends. Lol.
WAY TO GO GREG!!!
Thanks Dan!
Please make a video for plugin of Mia Laboratories Pi&Phi ✌✌
Dan, do you have Meldaproduction's MXXX plugin?
I fell in love with Alesis' Quadraverb (quality check seal from 1989) and if you want I could make a plugin just like this (GUI of the real hardware), including modulation options, within the MXXX frame. I wanted to do it for myself, anyway. The only catch is you'd have to have MXXX. (which I only have because I subscribed [to own!] Melda's excellent plugins)
The difference between modulated and non-modulated reverbs was rather noticeable on drums to me, cause I could hear how reverb tails sound in isolation.
And what's weird is that the modulated one's decay sounded smoother?
Hi Dan! In your channel membership, what's the difference between lowest and second lowest tier? And can you make a tutorial on how to create impulses inside the Daw? Like not a reverb but an impulse I already have+EQ so I can import it directly in let's say BiasFx2?
Afair no difference. People requested an in between tier, but I couldn't think of a way to differentiate it. If you want to EQ an existing impulse, just EQ it: bounce it through the setting you want, then load the result into your convolution plugin. As long as it's linear it should work.
@@DanWorrall thanks Dan I love you
Strange how it feels a bit fake when it turns out to be impulse responses, even if they sound perfectly fine. I have some very nice reverbs, but in practice I always grab a simple one first, otherwise I'll be twisting knobs for the following half-hour looking for perfection. A set of quality go-to impulse responses might just be a great thing to have.
The Arturia Lexicon is great!
Honestly if you missed those points, imagine the rest of us
Hola Dan, tus videos son muy utiles y educativos, los subtitulos me ayudan a entenderte mejor, podrias hablar de los plugins de Airwindows? Creo que podria ser interesante para todos. Saludos y gracias por tantos videos 🙏🏻
Would love an analysis of the MB MixHead plugin, as it's supposedly based on the secret sauce of Serban Ghenea
I wonder if they put some kind of bit-reduction on the input of that thing....or sample rate limiting. Like 8bit or 20k sample rate.
Point being, the original unit is probably pretty limited in that regard.
I didn't notice any bit crushing type distortion, but the limited bandwidth would be perfectly captured with an IR.
Can't remember exactly, as I've briefly played with a real PCM60 maybe 25 year ago and I haven't done any dissection of its internal algos. Even if the modulation will be there (I won't bet on it), it will be very subtle, not anything remotely similar to higher end units, where you can get its famous modulated tails. Also back then in guitar rigs those more affordable Lexicons were often combined with modulated delays, to make it like rich sounding (and more expensive) PCM81 and 91 units.
Can you look at any any of the Safari Pedal plugins?
Hey Dan, sorry for comment on wrong video :) but i have question about ableton group tracks, why reverb as a send on group track less effective
Less effective than what? I don't understand the question.
Hi Dan, i have a very interesting topic so like it would make for a great video if such thing exists
Here's the setup I'm aiming for:
- EQ Instance One: Placed before the input of Amp 1.
- EQ Instance Two: Sidechains the sound of Amp 2 to EQ Instance One and EQ Instance Three.
- EQ Instance Three: Placed after Amp 1 and sends feedback to EQ Instance One to match the sound of Amp 1 to Amp 2.
I need an EQ VST that can handle the following process:
EQ Instance One should iteratively adjust the EQ settings to account for the distortion and unique character of Amp 1, ensuring that the output sound of Amp 1 matches the target sound of Amp 2 as closely as possible.
Disclaimer: I'm not looking for an amp sim with tone match.
Do you know anything that fits these requirements? Or maybe another way of doing such a thing without spending weeks trying to recreate a tone? Im really stuck and frustrated :( haha
So, when it comes to this being worth it, I have to take into account it’ll be $30 at some point in the future. I’m currently happy with my reverb options, but I can see the value.
Yeah, the price on the web page doesn't mean much.
Another great video, thanks, just a small aside, given the impossibility of generating random numbers with a computer, and thus random noise within a reverb tail. To a programmer, it is entirely possible to use a random number generator, without reseeding, and getting an identical effect every run. It is by all accounts still random. Beware though, in the cave that is random number generation, there be Dragons!
I did think of that, so I tried bouncing files instead of nulling two instances: still nulled.
@@DanWorrall But the same code could be run, one version seeding every time and the other always using the same seed, both would sound the same to our mortal ears, but the one that is seeded would use up the computers entropy pool, if you want it to really be random that is, or as close to as is possible. Most sane programmers would I think, in this case, just use the same seed each time. The version that used the same seed would null.
@@DanWorrall The most basic example is the function rand() in the C standard library, if you dont seed it with the time, it will produce the exact same set of numbers every time.
heh, I suppose we could put this down to a convoluted use of a random number generator!
You could test for that by bouncing the test files at different timecodes. But I'm pretty satisfied it's just IRs now, and there's not even any pseudo randomness.
Ill keep the valhalla thx
I am interested in getting the BM60, but I will wait for a good sale. :)
And Dan, I don't know why you are not making big bucks doing voice/narration work. Or are you? :)
I'm pretty sure FabFilter compensate him well for his time when he does their tutorials for them. ;)
Great video as usual.
I’ve wondered whether the A/D stage would have any impact on the all pass filter network to create the reverb and whether IR’s are suitable for this, or if there is some kind of trade off. Presumably the IR is capturing some linear aspects of this stage, but it’s not possible to sample the reverb without it. I’m also not sure whether the plugins own input saturation is going to work as the hardware would.
I find IR’s to end up sounding quite stuffy in a mix, particularly if a lot of sources are feeding it. For individual tracks or certain uses, it’s not an issue, but some modulation or pitch change can add some excitement.
Valhalla Vintage and the various 224 plugins should share some similarities with the PCM60 algorithms, although they’d be harder to dial to the specific PCM60 settings. IIRC Avid Revibe II has some PCM60 inspired presets too.
IRs would include the filters in the AD/DA converters, but not any non linearities. The input distortion in the BM60 must be separate code as IRs are linear.
I wonder then what is under the hood of the relab lx480
I do NOT think the PCM60 has "the" Lexicon sound. So it would not surprise me at all if they don't have that tail modulation.
Edit: And yeah, both reverb A and B sounds to me like a bunch of densely packed echoes, just like the PCM60 does. And the MPX, and LXP etc. While later PCMs, and the 224 and 480 etc, there the tails are just one smooth continuous sound.
I'm wondering if there might still be modulation happening and that different instances of the same plugin might simply be pulling from the same seed at the same time, so they'd effectively have the same randomness applied to them. That way they should still null perfectly.
This is just an idea though, and I could be entirely wrong with this.
i thought reverb b blended the wet and dry better than reverb a where they sounded less inter-related
Can you create the modulation effect in reapers reaverb by modulating some of the parameters?
No. What parameters would you modulate? Closest you could get would be some kind of chorus or phaser type processing for the wet only signal, but it's not really the same.
So that we don't have to. Thanks!
I swear reverb A was a bit darker. But they do sound the same. Wouldn't care if I used the IR.
wonder how you feel about arturia’s lexicon emulation - it’s an emulation of the 224 and not the pcm 60, however. I never used any of the original hardware but it definitely has a flavor I don’t get from the valhalla reverbs (which also get a lot of use from me)
I might pick up the bettermaker once it goes on sale for lower than the current loyalty pricing of 70 bucks. if it hits 30 I could see it being a worthy impulse buy, no pun intended.
Look I love arturia stuff. But that reverb did not win me over. I have the UAD 224 and it IS the 224. A direct port of the algorithm and modeled i/o stages. The arturia is new and has some mod cons, but I couldn't get it to sound as good. That could be me, but I think why bother with an emulation when the realm thing is already available?
If you want to get Valhalla to sound closer to Lexicon 224, pick Dirty Hall or Dirty Plate modes with 1970 color and put bass multiplier at x1.00, also relax some high frequency damping.
I believe the famous Lexichip (with reverb modulation) came in the units after, and the last Lexichip unit was the MPX-1. However, the Lexichip was the same between units, but the PCB routing and extra chips differed. Such as the MPX-1 has a full multi effects chip as well.
The Bettermaker is a 1Gb download. False marketing imo. I'll probably pass even when it is $19 or $29 in the future. Have enough reverbs already...
Reverb B was more open sounding.Right?
Does it matter… well - back in the day sources were summed going into the reverb so it’s not like there’s the possibility that multiple reverbs at the same time have modulation interferences wasn’t really a thing either. So… nope
With a name like BM-60, I’d think there’d be a button that engages a 160db boost at 3 hz.
what if they are round-robining IR responses? That could mess up your null test.
If the original unit nulls it's self, then it's IR's should null too, even if multiple were taken. So it's highly unlikely they used multiple IRs in this scenario (as it would be pointless), and would only use them in a scenario where the original unit didn't null with it's self anyway. So the fact of it not nulling because of round robin, is telling you the original unit did not null either, and is exactly the behaviour you're trying to tell apart from, by doing the null test.
Just a note: When switching between two audio files/sources in an A/B comparison, human sensitivity to discerning a difference will be better if you flip back and forth in the middle of static audio information and not at the beginnings of measures or phrases (as most people seem to want to do). Since there tend to be transients at the downbeats and at the beginnings of measures, (such as voice and instrument onsets and offsets), those sounds can hide the subtle acoustic differences that the human brain might be able to otherwise distinguish if the switching is done instead in the middle of measures, notes, and phrases where the audio information is being held more constant without those transients. The acoustic psychophysics of this phenomenon are well characterized in the research literature. Cheers.
You're absolutely correct. But you could argue that if the differences are small enough to require that, they won't matter in a real mix.
@@DanWorrall But often the baseline question is whether they can be perceived at all. A real mix may have times when they can be discerned and times when they can't be. Ruling out differences during times that are more difficult to discern leaves the possibility that discernable differences still exist even in a real mix. Great music and videos. Sending appreciation from San Diego. ☀️👍🏻
I did this with ocean way so I could be more portable. Is there a difference? Sure. Is it slight as hell? Yep.
But it looks like it should sound like you remember… 😂
Although I looked after my outbound much better than BM seem to think one does/did…
Wouldn’t modulation still null if you could align the LFOs?
it would as long as there's no additional randomness going on, then it likely wouldn't (unless the pseudo-random 'seed', ie. starting point, is also the same and triggered at exactly the same time).
It doesn't have 128 IR samples, only like 6
Fascinating….just get altiverb and your done with reverb.
The coveted 90s reverb tone would be better - from the PCM 80 (or 81) & the PCM 90 (or 91).
Why didn't these plugin makers do a 1:1 code of it. I hear Lexicon & Izotope merged these days.
A bit afraid to comment on a Dan video, but if the plugin used time code as a random seed it could use modulation and still pass a null test.
Don't be scared, that's a very valid point. I did test renders as well just in case. But I'm hearing that the PCM60 didn't use modulation anyway, so I'm pretty sure that wasn't happening here.
@@DanWorrall makes sense. Thanks Dan :)
can we please have your favourite plate impulse 🙂
- so everything is IR?
- it always has been
me pretending i understand these 2 videos well enough to understand why youd be embarrassed
One point that I'd like to add is that I personally agree with White Sea's rants about analog-simulated UIs. I hate having a fake rack unit on my screen with useless buttons disposal. I love Valhalla UI, althought I also don't like their crazy planet names of the algorithms. Having said that, the original PCM analog UI is very fast and well done. Maybe a simpler interface, with the same buttons and names would do the job with less screen real state.
Why do I feel like I’m in Hyrule Castle?
yeh would be nice if they just admitted they are integrating ir technology its not even the worst selling point I would buy a half normal plugin and half ir plugin
Maybe they meant logarithmic lol!
My PCM60 short reverb is unbalanced like your software IR! Always annoyed me.
👌
For me B sounds better
Frankly I'm embarrassed that embarrassed is spelled "embarassed" in the title card. How embarrassing!
i will always endorse piracy
Only when it comes to actual big corporations
Plugin companies aren't big enough
2nd
Why are you embarrassed? It's late and I couldn't be assed watrching the video tbh
🫶
The gloomy music makes you sound so defeated over all this...
3rd