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Old 17th November 2009
  #91
Lives for gear
 

Quote:
Originally Posted by Casey View Post
Yes, Sean has a winner here!

I have been listening for the last week to a new (beta) native plugin from a developer in Europe. It absolutely nails the 480 sound (as opposed to the 960 or 96 sound). I really like it. Should drop before the end of the year. No idea on pricing, but probably much less than $1899.

Please read this carefully, this is NOT a Bricasti product, nor do I have anything to do with the company! stike

I love the resurgence of new reverbs. It will be interesting to see what TC does with their new native line. Harman and TC have always marched in lockstep. It will be interesting to see what new native developments come out of Denmark.



-Casey
Casey, this is indeed good news. Native compressors have upped the ante of late and now reverbs are coming to the party. All we need now are some really slick EQ's and Delay's with that classic analog sound and the ducking feature.

I still got to get my hands on a M7 with the new software to try out.

For now though I'm in happy land as two weeks ago a friend came to visit from a neighboring country bearing gifts. He had picked up a PCM 70 ver 2.0 for less than $300 in 2nd hand store. And he gave it to me... awesome friend I know.
Old 17th November 2009
  #92
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Casey's Avatar
 

Quote:
Originally Posted by Sonic Nomad View Post
He had picked up a PCM 70 ver 2.0 for less than $300 in 2nd hand store. And he gave it to me... awesome friend I know.
Very nice!



-Casey
Old 17th November 2009
  #93
Lives for gear
 

Am I alone in thinking the TC M30 Reverb sounded like 300 ping pong balls dropped onto a kitchen floor?

I was expecting a lot more to be honest. As for it sounding like the Eventide reverbs....not a chance. It sounded over bright, grainy and just not to my liking at all :(

Do the hardware units sound like this as I have never used a TC hardware reverb before only Eventide and Lexicon Pcm's?
Old 17th November 2009
  #94
Lives for gear
 
noiseflaw's Avatar
 

Quote:
Originally Posted by Captain Proton View Post
As for it sounding like the Eventide reverbs....not a chance.
He was referring to EOS not the TC M30.

Quote:
Originally Posted by Sonic Nomad View Post
Got to agree the samples sound good. It reminds me of the Eventide sound and knowing that Casey from Bricasti has commended the algo developer, it's on my short list.
That kinda silk for that kinda price - excellent!
Old 17th November 2009
  #95
Lives for gear
 

Aahh my fault, must learn to read properly and not skim

Looking forward to the new native reverb then.
Old 17th November 2009
  #96
Lives for gear
Just tried it on backing vocals and it sounds great!
Old 17th November 2009
  #97
ValhallaDSP
 
seancostello's Avatar
 

Quote:
Originally Posted by Sonic Nomad View Post
Sean, I only just realized it was you who replied.

it's scary what results some reverb plug-ins give.
Also nice for checking out impulses.
I wouldn't worry too much about the phase response at a given frequency. What you need to look for is if the phase response is uneven over the whole frequency range. A real room might show a VERY pronounced left or right lean for a given sine wave, but shift the frequency of the sine wave by a few hertz, and the lean should jump around a lot. What is critical is the density of the resonances and antiresonances, and that they are effectively randomly distributed.

Quote:
PS: I can't wait to get my hands on the Eos plug which will hopefully be soon.
Are you a fan of the Eventide verbs?
I've only heard the SP2016 ports to VST that Eventide / Princeton Digital did a few years back. Those sounded nice, but weren't an influence on my algorithms. Chris and Adam at Audio Damage are both big Eventide fans, and it was their tastes that shaped the development of the algorithms.
Old 17th November 2009
  #98
Lives for gear
 
Suda Badri's Avatar
 

Quote:
Originally Posted by divine source View Post
the difference is the M40 got room and plate algorithm as well.
So they are giving a reverb without a room or a plate?? buzzy... I am getting it now...
Old 17th November 2009
  #99
SEB
Lives for gear
 

I`ve been comparing the EOS with the UAD 250 which I think sounds very very good, better then the 250 in my TC6k. After a little while of tweaking the EOS I had it sound very similar to the 250. Close enough so that I`ve could have used both in my mix.

Good algo Sean!
Old 17th November 2009
  #100
Gear Nut
 

I really like this reverb. Smooth and open sounding, bright but not harsh. nice on vocals. Not very versatile but it does one thing and it does it well.

just did a quick a/b with the waves rverb and that just sounded like crap compared to this one. otoh, i don't own any of the fancy reverbs mentioned in this thread.

thanks tc.
Old 17th November 2009
  #101
Lives for gear
Quote:
Originally Posted by Patrice Baumel View Post
Not very versatile but it does one thing and it does it well.
That's my take as well. It compares with any of my current plugs and will get used. Great for free. thumbsup
Old 18th November 2009
  #102
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lozion's Avatar
 

Funny timing, I dl'ed the M30 in the hope of comparing it with my aging TC M-1 and just as I switched it on, the display froze and no signal passes...

Anyway, I like how TC verbs are unobtrusive on vocals and the M30 has that quality as well. It sits in the mix and isnt too upfront. I wish it had more controls but, yeah its a freebie. It can maybe now replace my dying (dead?) M-One.
Old 18th November 2009
  #103
Lives for gear
 
Farshad's Avatar
 

Quote:
Originally Posted by Animus View Post
Does anybody else see this as a clever preemptive strike to try and lessen the upcoming Lexicon plugin release?
+1 thumbsup
Old 18th November 2009
  #104
hmx
Here for the gear
 

TC M30 compared with Logic 9 Platinumverb

I felt motivated to do little comparision on logic 9 Macbook Pro OSX 10.5
with this TC M30, Platinumverb, space designer and ozon 4 reverb, see attached file. (logic 9 file)
IMO The M30 sounds close to what u can do with Logics Platinumverb.
Attached Files
File Type: zip HALL test.logic.zip (357.3 KB, 47 views)
Old 18th November 2009
  #105
Lives for gear
on SONAR 8.5 64-Bit on a Windows 7 64-bit PC this ran like crap - until I set the pluggin to run in the J-bridge wrapper. Runs great now and sound better than my convolution verbs.
Old 19th November 2009
  #106
Gear Maniac
Quote:
Originally Posted by Sonic Nomad View Post
All we need now are some really slick EQ's and Delay's with that classic analog sound and the ducking feature.
Soundtoys echoboy is what you're looking for.
Best delay plug out there in my opinion.

Cheers
Old 19th November 2009
  #107
Lives for gear
 
Coyoteous's Avatar
 

Certainly there are other problems, but this was added to the download page:

Quote:
Note for Snow Leopard users!
We've experienced a problem with some systems running Snow Leopard.
A preset change will lead to noisy output, and we recommend not to use the plug-in at present time.
We'll look into the bug and get it fixed. To make sure you get the fixed version, sign up for the free version, and we'll send you a mail when the fix is ready.
Old 19th November 2009
  #108
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DSpec1's Avatar
 

Quote:
Originally Posted by Coyoteous View Post
Certainly there are other problems, but this was added to the download page:
Yep, had this nasty issue last night....
Old 19th November 2009
  #109
Lives for gear
 

Quote:
Originally Posted by sprawl View Post
Soundtoys echoboy is what you're looking for.
Best delay plug out there in my opinion.

Cheers
Have the SoundToys Bundle and while they are great I'd still like a plug that can do a nice TC2290 emu and one the can deliver the quality pitch shifting delays that the Eventide Ultrahamonizers can deliver - then we are talking.
Old 19th November 2009
  #110
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Quote:
Originally Posted by seancostello View Post
I wouldn't worry too much about the phase response at a given frequency. What you need to look for is if the phase response is uneven over the whole frequency range. A real room might show a VERY pronounced left or right lean for a given sine wave, but shift the frequency of the sine wave by a few hertz, and the lean should jump around a lot. What is critical is the density of the resonances and antiresonances, and that they are effectively randomly distributed.
Ok some of that went straight over...

But hear you. I did notice uneven, odd and random behavior of the phases during sweeps that I tried on several verbs.
One some plugs specifically DSP based this is true. Random distribution? Interesting. I would have thought the results would be precise calculations in the case of a room, plate, hall or any other real world space emulation. Not so much in a special effect where one wanted an amount of the un-predicated. Or is this the norm with verb plug-ins? The use of random generators to create pseudo fluctuations to mimic a sense of distribution instead of a more accurate and obviously very complex model that would deliver a more realistic emulation.

Quote:
Originally Posted by seancostello View Post
I've only heard the SP2016 ports to VST that Eventide / Princeton Digital did a few years back. Those sounded nice, but weren't an influence on my algorithms. Chris and Adam at Audio Damage are both big Eventide fans, and it was their tastes that shaped the development of the algorithms
Sean, I got the Eos today and got to say WOW!!! Chris and Adam did well in their "shaping" of the algos.
I had an H8000 FW until recently (had to let her go as I closed the studio) and apart from the plethora of great effects on the unit some of the larger mod verbs where it's at for me. The SP2016 is very very different to some of these verbs found on the bigger Eventides.
There are two things that have softened the impact of parting with the H8000 in the last few weeks. They are the PCM70 ver 2.0 that was given to me and your Hall algo in Eos.
Well done Sean!!!
Old 19th November 2009
  #111
Lives for gear
 
fossaree's Avatar
I've got mine !

Have already used it on a recent project .

I would compare it favorably with my M3000 hardware .

;-)

For free , ain't got nothing to complain , actually !


(EDIT: Just realized I got the number of the beast on my number of posts ... I can hear Bruce Dickinson screaming on my ears !)

Last edited by fossaree; 19th November 2009 at 09:27 PM.. Reason: joke, for fun ;-)
Old 19th November 2009
  #112
Lives for gear
 
DSpec1's Avatar
 

I think it sounds beautiful -- when it doesn't blow my ears out...ah, TC will fix it and I will love it A LOT.
Old 19th November 2009
  #113
Gear Head
 
doktahyde's Avatar
 

Thank you
Old 19th November 2009
  #114
ValhallaDSP
 
seancostello's Avatar
 

Quote:
Originally Posted by Sonic Nomad View Post
One some plugs specifically DSP based this is true. Random distribution? Interesting. I would have thought the results would be precise calculations in the case of a room, plate, hall or any other real world space emulation. Not so much in a special effect where one wanted an amount of the un-predicated. Or is this the norm with verb plug-ins?
Most of the "hall," "plate," "room," etc. algorithms are very loose models of the physical processes, at best. Some of the high-end algorithms will model the early reflections of a space, but this only covers the first few hundred milliseconds. At that point, pretty much any acoustic space will have developed such a high echo density, that the echo response is essentially a random process. It is easier to view this as statistical, rather than deterministic. Instead of modeling the 20th order reflections, it makes more sense to say "there is x echo density, which resembles this random process, and the following frequency distribution..."

There are a few different approaches to reverb algorithm design. One is a physical model, where you adjust the walls, the shape of the room, the materials of the walls, etc., calculate the resulting echo pattern, and map it to your algorithm. This approach is probably best embodied by the high end algorithms in the TC6000, where great care is taken to model the early reflections of the space, and smoothly transition this into a late reverberation process.

An alternate approach looks at reverberation from a perceptual approach. The goal here is not to approximate a given space's physics, but to generate a sonic impression that resembles that space, or perhaps an ideal space. The Lexicon algorithms tend to lean towards a perceptual approach . Most of the Lexicon algorithms do not have separate early and late reverb, but model this as a single process. The 480L algorithms have "Shape" and "Spread" controls, that allow you to tailor the initial attack and decay of the reverb process. These controls can be tweaked to get things into a state that is very unlike most real-world reverberation processes, but are very useful in adding good sounding ambiance to a recorded signal.

Both of these approaches take huge short cuts. At some point, the directly modeled reflections are mixed with a late reverb process that uses delay lines feeding back on each other and themselves. The total amount of delay memory used is usually a small fraction of what would be needed to accurately model the statistics found in a real hall. The randomizers (modulators) found in these algorithms help to generate the impression of denser resonances, as the resonance density is directly linked to the total delay length.

A bit more about the "resonances" I refer to: You could model a room as a whole bunch of resonant bandpass filters in parallel. The peaks are the "resonances," while the valleys in between the peaks are the "antiresonances," or "notches" or whatever. In order to get even close to the sound of a reverb, you need to use a few thousand of these, distributed across the audible frequency range. A real hall can be viewed as having several BILLION of these resonances. Clearly, you can't do this in real time on any computer known, so you have to fake it.

A cheap way of generating a lot of resonances really quick is to use a delay line, and feeding the output back into the input. This will generate as many resonances as there are samples of delay, which is why this sort of feedback delay is referred to as a "comb filter" - the frequency response looks like the teeth of a comb.

However, the resonances in a standard delay based comb filter have a problem, in that they are regularly spaced. The ear picks up on this regular spacing right away, and attributes a pitch to this. A room or hall has a VERY random distribution of these resonances, where it is very difficult to impossible to hear any pattern in these resonances. So, the trick in artificial reverb algorithms is to use a bunch of feedback delay lines, while not having any regular patterns between the resonances that result in comb filtering being heard. This is the sort of random response I was referring to earlier - ideally, you will have several resonances per Hz, with a different random pattern between left and right channels.

The time-varying randomizers / modulators are usually used to take an algorithm that has a resonance every few hertz, and make it sound like it has several randomly distributed resonances per Hertz. However, the modulation by itself can't fix an algorithm that has regularly spaced resonances. You need to design the algorithm in such a way that the basic resonances are as randomly spaced as possible, with no audible clusters of peaks or big holes in the distribution.

Another problem with reverbs, both artificial and real-world, is having one or more of these abstract resonances being much louder than the other ones. This will result in a ringing sound, or a beating sound if two or more of the extra-loud resonances are next to each other.

Quote:
The use of random generators to create pseudo fluctuations to mimic a sense of distribution instead of a more accurate and obviously very complex model that would deliver a more realistic emulation.
The real world has randomization going on, due to the slight changes in the speed of sound generated by temperature differentials. But, yeah, the randomization was originally used to increase the number of "perceived" resonances.

Quote:
Sean, I got the Eos today and got to say WOW!!! Chris and Adam did well in their "shaping" of the algos.
Thanks! I'll pass this on to the Audio Damage guys.
Old 20th November 2009
  #115
Lives for gear
 

Quote:
Originally Posted by seancostello View Post
Most of the "hall," "plate," "room," etc. algorithms are very loose models of the physical processes, at best. Some of the high-end algorithms will model the early reflections of a space, but this only covers the first few hundred milliseconds. At that point, pretty much any acoustic space will have developed such a high echo density, that the echo response is essentially a random process. It is easier to view this as statistical, rather than deterministic. Instead of modeling the 20th order reflections, it makes more sense to say "there is x echo density, which resembles this random process, and the following frequency distribution..."

There are a few different approaches to reverb algorithm design. One is a physical model, where you adjust the walls, the shape of the room, the materials of the walls, etc., calculate the resulting echo pattern, and map it to your algorithm. This approach is probably best embodied by the high end algorithms in the TC6000, where great care is taken to model the early reflections of the space, and smoothly transition this into a late reverberation process.

An alternate approach looks at reverberation from a perceptual approach. The goal here is not to approximate a given space's physics, but to generate a sonic impression that resembles that space, or perhaps an ideal space. The Lexicon algorithms tend to lean towards a perceptual approach . Most of the Lexicon algorithms do not have separate early and late reverb, but model this as a single process. The 480L algorithms have "Shape" and "Spread" controls, that allow you to tailor the initial attack and decay of the reverb process. These controls can be tweaked to get things into a state that is very unlike most real-world reverberation processes, but are very useful in adding good sounding ambiance to a recorded signal.

Both of these approaches take huge short cuts. At some point, the directly modeled reflections are mixed with a late reverb process that uses delay lines feeding back on each other and themselves. The total amount of delay memory used is usually a small fraction of what would be needed to accurately model the statistics found in a real hall. The randomizers (modulators) found in these algorithms help to generate the impression of denser resonances, as the resonance density is directly linked to the total delay length.

A bit more about the "resonances" I refer to: You could model a room as a whole bunch of resonant bandpass filters in parallel. The peaks are the "resonances," while the valleys in between the peaks are the "antiresonances," or "notches" or whatever. In order to get even close to the sound of a reverb, you need to use a few thousand of these, distributed across the audible frequency range. A real hall can be viewed as having several BILLION of these resonances. Clearly, you can't do this in real time on any computer known, so you have to fake it.

A cheap way of generating a lot of resonances really quick is to use a delay line, and feeding the output back into the input. This will generate as many resonances as there are samples of delay, which is why this sort of feedback delay is referred to as a "comb filter" - the frequency response looks like the teeth of a comb.

However, the resonances in a standard delay based comb filter have a problem, in that they are regularly spaced. The ear picks up on this regular spacing right away, and attributes a pitch to this. A room or hall has a VERY random distribution of these resonances, where it is very difficult to impossible to hear any pattern in these resonances. So, the trick in artificial reverb algorithms is to use a bunch of feedback delay lines, while not having any regular patterns between the resonances that result in comb filtering being heard. This is the sort of random response I was referring to earlier - ideally, you will have several resonances per Hz, with a different random pattern between left and right channels.

The time-varying randomizers / modulators are usually used to take an algorithm that has a resonance every few hertz, and make it sound like it has several randomly distributed resonances per Hertz. However, the modulation by itself can't fix an algorithm that has regularly spaced resonances. You need to design the algorithm in such a way that the basic resonances are as randomly spaced as possible, with no audible clusters of peaks or big holes in the distribution.

Another problem with reverbs, both artificial and real-world, is having one or more of these abstract resonances being much louder than the other ones. This will result in a ringing sound, or a beating sound if two or more of the extra-loud resonances are next to each other.

The real world has randomization going on, due to the slight changes in the speed of sound generated by temperature differentials. But, yeah, the randomization was originally used to increase the number of "perceived" resonances.

Thanks! I'll pass this on to the Audio Damage guys.
Sean, thanks for the detailed response. Some really interesting stuff.
This not only explains a lot but also helps me and hopefully others understand better how these current processes work.

At times I play around with the Quantum FX workbench and all this random/modulation talk has inspired me to mess with just that but not with something as complex as a verb but probably try make a delay with a saturator and basic random/modulator in the feedback chain.
Quantum FX is great for non programmers like myself to play around and create plugs and have some fun
Old 21st November 2009
  #116
Lives for gear
 

Quote:
Originally Posted by Casey View Post
I have been listening for the last week to a new (beta) native plugin from a developer in Europe. It absolutely nails the 480 sound (as opposed to the 960 or 96 sound). I really like it. Should drop before the end of the year. No idea on pricing, but probably much less than $1899.

...
is it this new plugin you are talking about?

KVR: KVR Developer Challenge 2009

HybridReverb

http://www2.ika.rub.de/publications/2009/borss09vst.pdf
Old 21st November 2009
  #117
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fossaree's Avatar
Quote:
Originally Posted by audio ergo sum View Post
This is one seems to be quite good , although there's no AU :-( ...
Old 21st November 2009
  #118
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jamwerks's Avatar
 

Quote:
Originally Posted by seancostello View Post
A real hall can be viewed as having several BILLION of these resonances. Clearly, you can't do this in real time on any computer known, so you have to fake it.
Enjoyed reading your informative post! thumbsup

Question: do convolution reverbs using IR's achieve this "Billion" or close? Do they give you the "real" room? What are the limitations of IR?

Thanks in advance,
Old 21st November 2009
  #119
Gear Addict
 
lostinmusic's Avatar
 

I recommend anyone reading this to demo the Redline Reverb. To my ears it's the nearest I've heard to a good hardware reverb yet. This TC freebie is just that, a freebie. I'll definitely be demoing whatever this new 480 style plugin turns out to be though..
Old 21st November 2009
  #120
ValhallaDSP
 
seancostello's Avatar
 

Quote:
Originally Posted by jamwerks View Post
Question: do convolution reverbs using IR's achieve this "Billion" or close? Do they give you the "real" room? What are the limitations of IR?
Convolution reverbs don't achieve the billions of resonances. Technically, a convolution impulse is an FIR filter, which has NO resonances (i.e. poles) - just anti-resonances (i.e. zeros). In practice, this makes no real difference. Convolution can have a high enough perceived resonance density to accurately simulate a non-time varying room.

However, most rooms are time-varying, due to temperature variations in the room which cause small variations in the speed of sound. These small variations are enough to cause largish errors in the late frequency response of the convolution impulse, and the small pitch changes that the real room would have cannot be captured by convolution. A real acoustic space will demonstrate the same types of randomization as found in the best algorithmic reverbs, although a real acoustic space will have a far greater number of modulation sources, with far more subtle "settings" of each modulator.

Having said this, in a fairly small acoustic space (a "room," not a "hall"), the speed of sound differences will be fairly subtle. I think that using convolution to model smaller spaces is a good idea. This also works well with how the cost of convolution scales with the length of the impulse - long halls will take more CPU than short rooms. Compare this to a typical algorithmic reverb, where the CPU is the same for any decay length.

A counter to the argument I just gave: People love the EMT250 for short ambiences. This algorithm is TOTALLY time varying.

Getting back to the billions of resonances: The billions of resonances are from a physical perspective. From a perceptual perspective, it is highly unlikely that we can perceive those billions of resonances accurately, or discriminate them from several hundred thousand resonances with the right type of time variation applied. I haven't seen many studies on this, but one paper suggested that a few thousand resonances can do the trick, if time variation is properly applied to each resonance separately. This approach (a few thousand parallel 2nd order filters, each with its own randomizer) is still too expensive for modern computers, although it would lend itself well to GPUs. I would want to hear the results before commenting on the quality of this approach. Many academic papers have made claims about how a given reverb approach is indistinguishable from an acoustic space, but very few of those claims hold up over time. The demands of the marketplace put higher demands on sonic quality than what seems to be going on in acadamia.
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