Reverb Subculture

[seancostello]

Originally Posted by Nobody Special

Since you've been talking about allpasses, I'll pass along an observation. Although the traditional view is that allpasses increase reverb density, that's only true up to a point. The number of reflections per second is still limited by the Nyquist theorum (and probably by the auditory cortex as well). With well-chosen allpass values, that limit can be reached quite rapidly. After that point, the significant effects are phase and frequency.

This is an interesting point. We have been discussing "constant density" algorithms in this thread. If the initial echo density approaches the maximum density supported by the Nyquist theorem, or the maximum density that can be perceived by our ears, then having an echo density that increases over time is a moot point.

From the discussion of the EMT-esque algorithm, and listening to older reverberators, it appears that there were many algorithms that took such a constant density approach. Some of these seem to be improved Schroeder algorithms, others undoubtedly use far different algorithms - there are a lot of ways to get density quickly.

One of the big problems with any algorithmic reverberator, be it constant density or increasing with time, is the ability of the ears to quicky pick up repetitive patterns in seemingly random signals. Most delay based time invariant reverberators will have an audible repetition rate for long decays, where the repetition rate occurs at the sum of all of the delay lengths for a well tuned reverb (and at a much faster rate or series of rates for a poorly tuned reverb). The initial use of randomizers within reverbs seems to have been to reduce or eliminate this repetition artifact. Recent publications have made a convincing argument that real acoustic spaces are time varying, but my guess is that the early time-varying reverbs were not trying to emulate this at all.

The time variation found in most commercial reverbs can go far beyond what would be found in most real-world situations. Boston Symphony Hall may have randomization of the delay times, but an algorithm like the PCM70 Concert Hall, or the Eos Superhall, or the DP/Pro Expert Reverb displays artifacts that would probably not be present without some extensive modifications to that building, such as a series of active bonfires within the structure. This larger-than-life artifact has now become part and parcel of many styles of electronic and ambient music. The Lexicon manuals may warn against pitch warbling on piano with the Concert Hall algorithm, yet many people use this algorithm on piano for this very reason. The Brian Eno / Harold Budd albums display some pitch artifacts that I am now able to hear as part and parcel of the EMT250 used on those songs, and the music would not be nearly as cool without those sidebands decaying into the aether.

[seancostello]

I had a nice summer walk today, and was thinking about reverb, and why I think about reverbs. In some ways, it is like being obsessed with Sodoku or Go or chess or something like that - there are an endless variety of structures that can be mentally manipulated.

My question is for the people that have been posting on this thread, and build reverberators for a living, or at least as a serious hobby: How did you end up making reverberators? Was it through a EE route, or as a musician, or something else entirely?

[Froombosch]

When starting a recordingstudio I always had the idea I needed an Eventide processor. I bought myself an Eclipse and started building presets for them. I was very active with Eventidehelps and enjoyed the building of presets and the active community there. Finally I bought myself a H8000 and started constructing new presets. I build a few presets for this machine (Dimension D replica etc), that are now inside the H8000. Finally I got interested in reverb technology and am now working on some algos that will be ported over to VST world.

[Froombosch]

Anyone any thoughts on the reflections that are exist inside a wood? Anyone who has found a paper on this topic?

[zmix]

Originally Posted by seancostello

How did you end up making reverberators? Was it through a EE route, or as a musician, or something else entirely?

What a fun question, Sean. All of these are factors for me.

Growing up in Catholic school and listening to the priests (speaking and singing) and of course the pipe organ in the Cathedral. The rolling quality to that reverb is something I have seldom heard in an artificial reverb.

Much later, on an early visit to a studio hearing the tape stop during a mix and hearing just the reverb tail... this made a huge impression.

"Coming of age" in the technology obsessed late 1970s and the reverb drenched 1980s fueled the obsession with sculpting space. Getting an Electrical engineering degree was simply more gasoline on that fire.

Discovering object oriented DSP construction software which allowed me to quickly test theories and produce the 'echos in my head' that had eluded me until then.

[Nobody Special]

Originally Posted by seancostello

My question is for the people that have been posting on this thread, and build reverberators for a living, or at least as a serious hobby: How did you end up making reverberators? Was it through a EE route, or as a musician, or something else entirely?

Happened in a lot of ways for me. Childhood was a mix of science and music. I spent early adulthood on the road as a guitarist. Musical interests matured so I went back to school and got a couple of music degrees. During this time I fell into computing, having been invited to participate in computer music at MIT (and I wasn't even in school there). I've run a lot of new music concerts in all sorts of spaces and even conducted a few times (luckily stopping before anyone was seriously injured). Early engineering years included speech recognition, operating systems design, medical computing. When I came to Lexicon it was a good opportunity to put it all together. I was doing other things there for several years before I had an opportunity to work on the 'verbs and other FX. At this point it's a mix of science, engineering and intuition. And I'm still doing the new music concerts when I can. I suspect most of us are mongrels in one way or another.

[Nobody Special]

Originally Posted by seancostello

This is an interesting point. We have been discussing "constant density" algorithms in this thread. If the initial echo density approaches the maximum density supported by the Nyquist theorem, or the maximum density that can be perceived by our ears, then having an echo density that increases over time is a moot point.

There's still a lot of interesting stuff that goes on after you reach maximum density. Even though there are no more reflections to be had, the impulse response is still constantly changing as peaks and troughs line up in different combinations. As has been pointed out, there are still many hazards to face when going through the same old allpasses time after time. Part of the art is on knowing how to minimize those effects.

[seancostello]

Originally Posted by Nobody Special

There's still a lot of interesting stuff that goes on after you reach maximum density. Even though there are no more reflections to be had, the impulse response is still constantly changing as peaks and troughs line up in different combinations. As has been pointed out, there are still many hazards to face when going through the same old allpasses time after time. Part of the art is on knowing how to minimize those effects.

The "constant density" algorithms seem to take the solution of avoiding allpasses in recursive loops. If things are dense enough, why put allpasses in there?

[seancostello]

Originally Posted by Froombosch

Anyone any thoughts on the reflections that are exist inside a wood? Anyone who has found a paper on this topic?

AES E-Library: A Digital Reverberator Modeled After the Scattering of Acoustic Waves by Trees in a Forest

[Nobody Special]

Originally Posted by seancostello

The "constant density" algorithms seem to take the solution of avoiding allpasses in recursive loops. If things are dense enough, why put allpasses in there?

Because they're still doing the other good thing they do, which is to modify phase.

[zmix]

Originally Posted by Froombosch

I do not think so. The diffusers divide the reflections more after every feedback return. So this topology had increased amount of reflections.

OK here's a quick synopsis of the Dattorro "Expert Reverb" algo in the Ensoniq DP/Pro

Lo Cut frequency and gain parameters

Hi Cut frequency and gain parameters.

Pre Delay 0-100ms

6 series diffusors on each input.
6 left, 6 right, each adjustable from 0.00 to 23.0ms in 0.1ms increments.

Master Diffuser Size 0-100%
Diffusion amount scaler -100% to +100%




Reverb Tank

Master Reverb Size scaler 0-100%

Mid Decay: 0.00 (produces no output) to 1000s
Low Decay 0%-400%
Low-Mid Xover 3hz-20khz
HF Damping 3hz - 20khz

Density 0%-100% this controls diffusion within the reverb tank, so it's possible to create a "constant density" reverb by setting this parameter to 0%.

8 delays in the reverb tank each adjustable from 0.00ms - 200.0ms These delays are matrixed somehow and there is an "Expert Reverb 2" algo which has a different arrangement of these 8 delays.

4 modulation LFOs which modulate the delays in the reverb tank, each has a depth adjustment from 0.00ms to 200.0ms and a rate adjustment from 0.00hz to 1.760khz.


Quite a flexible algo.... Certainly called "Expert Reverb" for a good reason...

[seancostello]

Originally Posted by zmix

Density 0%-100% this controls diffusion within the reverb tank, so it's possible to create a "constant density" reverb by setting this parameter to 0%.

So, if density is set to zero, there is no increase in density whatsoever as the reverb decays?

8 delays in the reverb tank each adjustable from 0.00ms - 200.0ms These delays are matrixed somehow and there is an "Expert Reverb 2" algo which has a different arrangement of these 8 delays.

What types of delays are being adjusted? Allpass delays, parallel combs, FDN delay lines, output taps?

Certainly called "Expert Reverb" for a good reason...

For the people at Ensoniq who knew what the algorithm was like under the hood, having all the delay lengths available for adjustment at the tuning level would have been great for creating distinctive presets. For your average user, about 98% of the possible delay settings are going to sound really bad. I think Ensoniq must have had more trust in mankind than I do.

[zmix]

Originally Posted by seancostello

So, if density is set to zero, there is no increase in density whatsoever as the reverb decays?

Basically, when the density is a 0% it sounds like 8 nested delays. The "Expert Reverb 2" variation is simply a different matrix structure of these 8 delay lines.

What types of delays are being adjusted? Allpass delays, parallel combs, FDN delay lines, output taps?

I am not sure if the delays in the tank are allpass delays , but I suspect that they are. When the density parameter is increased the repeats become diffuse in proportion to the lengths.

For the people at Ensoniq who knew what the algorithm was like under the hood, having all the delay lengths available for adjustment at the tuning level would have been great for creating distinctive presets. For your average user, about 98% of the possible delay settings are going to sound really bad. I think Ensoniq must have had more trust in mankind than I do.

Precisely. I would go one step further and say that it is highly unlikely that the average user would create a useful reverb preset.

That's why people like me did the presets for the box...

[seancostello]

Originally Posted by zmix

Basically, when the density is a 0% it sounds like 8 nested delays. The "Expert Reverb 2" variation is simply a different matrix structure of these 8 delay lines.

By nested, do you think that these delays are parallel in an FDN, or parallel combs, or series allpasses, or...? If they are in an FDN, there will be some default echo increase with time, unless the scattering matrix is being changed by the Density parameter.

I am not sure if the delays in the tank are allpass delays , but I suspect that they are. When the density parameter is increased the repeats become diffuse in proportion to the lengths.

Sounds like there are allpasses in there, although it could be the variable scattering matrix. Gerzon showed such matrices as early as 1971, but they get kinda expensive for an 8-delay FDN.

Precisely. I would go one step further and say that it is highly unlikely that the average user would create a useful reverb preset.

That's why people like me did the presets for the box...

There should be some special code, or hardware key, needed to access these parameters. Something along the lines of "heyre be dragons." Maybe the two people turning keys at the same time that you need to launch a missle..

[Froombosch]

Originally Posted by seancostello

AES E-Library: A Digital Reverberator Modeled After the Scattering of Acoustic Waves by Trees in a Forest

Thanks! I will get a copy of that paper! :)

[seancostello]

Originally Posted by zmix

Discovering object oriented DSP construction software which allowed me to quickly test theories and produce the 'echos in my head' that had eluded me until then.

What software? I have played around with a bunch of different environments (MAX/MSP, Pd, Supercollider, Reaktor), and haven't found anything that really does what I need it to do, except for my own C++ modules.

As far as my own background, I am kind of a mutt. Studied at CCRMA as an undergraduate, but majored in Anthropology, so I would not describe myself as a "techie" by any stretch of the imagination. Did the Seattle guitar slacker thing, got obsessed with fuzz boxes, started putting together circuits. Later came techno, started playing with synths and trying to build them (unsuccessfully). Computer music seemed like a more fruitful path, so talked my way into a year long course at the University of Washington. Did really well from a technical perspective, for some unknown reason. Ended up getting a job in the Bay Area with some CCRMA PhDs, which led me down my current career path, as well as being a great way to learn on the job from some really smart guys.

I first worked with reverbs as it seemed like a good way of getting some complex sounds with minimum CPU cycles. I could matrix together 4 modulated delay lines with near-infinite gain, and get sounds almost identical to granular textures using 100 grains, at about 1/100th of the DSP of the granular processes. I tracked down every paper I could read on the subject. Many of them took me a few years to digest, but I think I get the gist now. 1998 and 1999 were spent learning the basics, the early 2000's were spent on developing different structures, and the late 2000's have been spent learning how to make the structures sound good.

[zmix]

Here are some examples of the Dattorro Algo in the form of the ENSONIQ DP/PRO "Expert Reverb".

These are some basic examples to illustrate the difference between the FDN of the Expert Reverb 1 and Expert reverb 2 variations.

In all examples the Expert Reverb 1 is mono panned left and the Expert reverb 2 is mono panned right.

The DP/Pro is a dual processor device so these are simultaneous yet uncompromised iterations.

In these examples you will hear the two reverbs with only the following parameters altered: "Diffusion" (Input diffusion) and "Density" (tank diffusion).

Diffusion 100% Density 100%
Diffusion 0% Density 100%
Diffusion 100% Density 0%
Diffusion 0% Density 0%


The input diffusors, normally 6 left and 6 right in the stereo routing configuration, are combined in the mono routing and are also identical in each algo.

In other words, in this "Dual Mono" routing configuration these "Left" and "Right" diffusor chains are combined in parallel and so are identical for both of the inputs.

"L"
4.2ms
5.7ms
13.1ms
18.5ms
23.4ms
20.0ms

"R"
16.2ms
9.9ms
6.0ms
8.0ms
10.0ms
12.0ms

The 8 Tank Delays are identical for each algo and set as follows:

47.0ms
97.0ms
181.0ms
77.7ms
141.0ms
61.0ms
177.0ms
31.0ms

Reverb time is set to 4.5 seconds with HF damping at 11khz.

PLEASE NOTE: The only difference between the reverb you hear on the left side and the one on the right is in the delay matrix routing in the reverb tank.

[zmix]

Here is a set of interesting images...

These are graphical depictions the pitch modulations produced by Program Button 1 "Small Concert Hall - B" and Program Button 5 "Percussion Plate - A" from the last software revision of the original 224, v4.x, both with the default "Mode Enhancement" value of "02" The depth of the pitch variation in the Concert Hall at this setting is approximately 6 cents.

The horizontal axis is divided into demarcations at the bottom of each image representing 1 second. The entire image represents 45 seconds.

The pitch is represented by the BLUE squiggles at the bottom of the images..... the Concert Hall is shown in stereo: Left on top Right on bottom.
The Plate is shown only with the left channel output. When the RT=0 there is NO modulation in the right output

[seancostello]

Originally Posted by zmix

These are graphical depictions the pitch modulations produced by Program Button 1 "Small Concert Hall - B" and Program Button 5 "Percussion Plate - A" from the last software revision of the original 224, v4.x, both with the default "Mode Enhancement" value of "02"

So, you have access to an original 224? Where are the sound files? I DEMAND access to these! Based on nothing more than an inborn sense of entitlement, that is increasingly common today.

In all seriousness, any sound files would be very cool. Not many people seem to have access to a non-XL 224.

[seancostello]

Originally Posted by zmix

Here is a set of interesting images...

These are graphical depictions the pitch modulations produced by Program Button 1 "Small Concert Hall - B" and Program Button 5 "Percussion Plate - A" from the last software revision of the original 224, v4.x, both with the default "Mode Enhancement" value of "02" The depth of the pitch variation in the Concert Hall at this setting is approximately 6 cents.

The horizontal axis is divided into demarcations at the bottom of each image representing 1 second. The entire image represents 45 seconds.

What part of the image shows the pitch variation? Could you explain in more detail what we are seeing? I had presumed that the bottom gray part showed the deviation from true pitch, but I might be incorrect.

[samiam]

You know, this is one of the best threads I have seen in a long time.

I, too, have long had an interest in reverberation. I have never been fully satisfied with the old-fashioned allpass filter + delay lines technique of reverberation; I never felt this made completely realistic reverbs, especially for medium sized spaces with a lot of reflections (such as a concrete stairwell).

One idea that comes to mind, after reading this thread, is to see if it is possible, with today's computational power, to use something fancier than an all-pass filter for diffusing the sound, such as using a convoluter with a very short impulse response convolution (mainly as an EQ and as a way to smear transients). However, while computing power has gotten a lot better, we may still not be at the point where it's practical to have a few concoluters, albeit short ones, in the core of a reverb algorithm.

While I have made some very satisfying music using the reverb in my old Roland SRV-2000 on guitar, this reverb was really an effect more than it was a realistic reverb. The SRV-2000 had, for me, a good sound, but it didn't really add color or flavor to the sound it was reverberating; it however had a very static reverb tail that is really nice for "infinite reverb" special effects.

I also have a PCM-90 with the dual-reverb card (in effect, a PCM-91). People have been talking about the "Concert Hall" algorithm from the 224 on this thread; I don't know how much the 90's "Concert Hall" sounds like the 224's Concert Hall, but the truth is that I have never been satisfied with its sound. I feel "Random Hall" sounds a lot better, but is a bit thin sounding for me with too much tendency for resonating standing waves.

The PCM-90's "Ambience" algorithm is excellent for getting a realistic sound for small to medium sized spaces (People who claim the Lexicons can't make a realistic space should listen to the PCM90's "Living Room" preset before passing judgment); however it doesn't work too well if one makes the decay too long; too much of a "looping" sound in the tail; the Impulse Responses I have heard from an old EMT 2XX digital reverb have the same problem.

This is the only realistic algorithm the 90 has; the others ("Random Hall", "Rich Plate", "Concert Hall", and the algorithms in the dual-reveb cart) are definitely "effect" reverbs.

I just downloaded for free off of the internet a program by Alelsey Vaneev called "Voxengo Impulse Modeler" which uses a form of ray tracing to create reverberant washes. What one does is design a room by placing walls with various reflective and absorbent characteristics in a virtual 2-dimensional space; one can also set the absorbent characteristics of air. The program then creates an impulse response convolution that can be used by any IR reverb by simulating the sound bouncing off of the reflective surfaces and being absorbed both by air and by walls to create a very realistic reverberant sound.

No modulations to stop ringing tails is needed (or in the case of some of the PCM-90s and Roland R-880 presets, we keep the ringing tails because that's better than modulation for some sources); the sound is a good deal more realistic since it more closely m odels what sound does when it reflects off of walls in a large hall.

Older versions of the program had problems with tails being really grainy. Newer versions of the program solve this problem by allowing the emitter (the point source that the impulse response emanates from in the virtual space) to emit up to 1,000,000 rays (the default of 40,000 rays is fast, but quite grainy). This, for most settings, results in a very deep, realistic wash being generated. If one tries to create a wash with a very long decay time (like the Quantec "Taj Majal" preset), there will be some grain at the very end of the trail; the way I work around this is to take the IR with the long tail, run the IR through the convolution reverb using itself as the convolution, and using the resulting 30-second long very thick wash as an IR for that kind of special effect.

One can adjust the size and diffusion of the space by adjusting only a single parameter. Decay isn't a single knob; one needs to change the damping characteristics of the air and walls to adjust the decay.

It takes about 20 minutes to an hour on my 2-year-old computer to generate a really good wash, but the result is amazing. I finally have a way of simulating the wash of things like concrete stairways using a simulated digital reverb.

The program is a fully-functional free download; the only thing disabled in the trial version is the ability to save one's own reverb design, and one can only create three impulses before restarting the program. One is free to use impulses created with the trial version for any purposes however, and registration is only $40 (considering I spent some $1500 for my PCM-90 about a decade ago, this is a bargain).

The attached .wav file is an impulse responses using the program's "Hall 2" preset; the only changes I have made is increasing the number of rays emitted up to a full 1,000,000 rays, increasing the length of the reverb generated to 16 seconds (by default, it only generates a 4-second reverb), and making the air, draperies, and walls more reflective. I then took the impulse response, opened it up in an editor, removed a single transient peak in the file, normalized the file (in a manner so it sounds balanced in both speakers), the converted it to a 16-bit 44.1 khz wave.

I'm really amazed one can get a sound that matches or exceeds the sound of the old-school reverbs in a program costing only $40.

I wonder how long until Moore's Law catches up so we get to the point where we can just have real-time ray-tracing reverbs to add to our mix. It's parallelizes very nicely; each ray can be calculated by a separate processor and mixed together at the final stage (or have like 50 DSPs, each one doing 20,000 rays or so in real time).

[Nobody Special]

Originally Posted by samiam

One idea that comes to mind, after reading this thread, is to see if it is possible, with today's computational power, to use something fancier than an all-pass filter for diffusing the sound, such as using a convoluter with a very short impulse response convolution (mainly as an EQ and as a way to smear transients). However, while computing power has gotten a lot better, we may still not be at the point where it's practical to have a few concoluters, albeit short ones, in the core of a reverb algorithm.

You're describing something similar to the 'Room' algorithm in the PCM96. There are 48 real-world impulse responses that can optionally be dumped into a standard reverb. The impulse responses don't use a full convolver. Instead, they're mathematically reduced into a form that allows some realtime modification. Unlike a convolver, the impulses can easily be rubberbanded in time, EQ'ed and reversed. While this algorithm was originally done with film post in mind, it has a lot of flexibility and potential for future growth. In a very real sense, it's the grandchild of the ambience algorithm that you like in the PCM90.

I have no wish to get into the convolution vs. algorithmic debate (well only a small wish). The best convolved reverbs sound excellent. They're just a little hard to match up with material sometimes. Algorithmic reverbs offer greater flexibility in terms of EQ and other important characteristics. For that reason I don't see them going anywhere.

Ray-tracing is similar. Your example sounded very good, but it didn't sound necessarily superior to a well-done algorithmic or convolved reverb. Even though Moore's law will indeed make processing power a moot point, I still think that --with the exception of early reflections -- ray-tracing may represent serious overkill. But stay with it and prove me wrong, eh?

N.S.

[zmix]

Here is something a little different, a simple Clave, long (10s) decay and 100ms predelay ... I snapped this photo with the camera in my cellphone 2 minutes ago...

[zmix]

Pretty basic, but this is the WET signal only. The real magic happens when this is heard with the direct signal... it's stunning.

Pretty much the stock setting except the decay is set to 6.5s

The second and third samples have the 'depth' slider at Minimum and Maximum.

[samiam]

Zmix, if you have a chance, could you post a comparison of the 224 "Concert Hall" with the PCM-90 "Concert Hall" or let me know what you think about how they compare? The PCM-90 "Concert Hall" was never my cup of tea and I'm wondering if I would feel the same about the original classic.

Concerning classical all-pass delay bank reverbs, I think they will have a place in music making for a long, long time, just as tubes will have a place in music production for a long time. It may not be the most accurate way to make a reverb, but it has a distinct sound that defines some genres of music. Trace is not trace without a Lexicon wash with a long decay time, for example. Another example of a great song dependent on the Lexicon sound is "12:18" by Global Communication from their album "76:14" (that sounds like "Random Hall" with a really long decay time to my ears). Convolution (and possibly eventually ray-tracing) reverb is not going to change that. And, no, Convolution can not properly emulate a hardware reverb because so much of hardware's sound depends on modulation and what not.

One idea I have is if there is a process where we can automate the creation of an allpass or whatever filter bank, where we tell the computer "OK, given this convolution, find me a bank of filters and delay lines that most closely resembles this sound. Make sure I can adjust the decay time and room size", wait all night and have a reasonable-sounding reverb algorithm that can be reasonably well tweaked. The devil is in the details, mainly, how will the computer know how much a given reverb algo sounds like the convolution we want it to sound like (once we have a good heuristic for that, we can use genetic-type algorithms to tweak the reverb tank until it sounds the way we want).

I love the sound of classical reverbs; indeed I posted seven years ago about why they will continue to survive even with convolution reverbs in existence:

usenet posting with Message-ID 8759d333.0210151903.7db31f0f@posting.google.com

Anyway, one of my dreams is to learn how to make a classical reverb; how to program an all pass filter, delay lines and what not. I can program in C and have programmed in assembly (anyone else remember the 6502), and am wondering where I can get information to learn how to do that.

Anyway, this is a great thread and I appreciate all of the people contributing!

- Sam

[seancostello]

Originally Posted by samiam

Anyway, one of my dreams is to learn how to make a classical reverb; how to program an all pass filter, delay lines and what not. I can program in C and have programmed in assembly (anyone else remember the 6502), and am wondering where I can get information to learn how to do that.

Start here:

http://www.stanford.edu/~dattorro/EffectDesignPart1.pdf

Then read this:

www.media.mit.edu/~billg/docs/ms-thesis.ps.gz

You should also check out the reverb info at:

Spin Semiconductor - Effects

These will give you a feel for the allpass loop style reverbs.

[samiam]

A real quick reply:

zmix: I have attached a sample of a TR-808 clave run through an IR generated by Alelsey Vaneev's Voxengo Impulse Modeler; I took the "Hall 2" preset and minimized the decay caused by air to get a reverb time around 10 seconds. It's not nearly as precise as a Lexicon to program; there is no "reverb time" setting.

What I had to do is modify the absorption properties of the walls and the air ("air" would be more adsorbent when, for example, simulating a concert hall full of people), test out the reverb until I got about a 10 second decay time, spend 20 to 30 minutes doing a final 1,000,000 ray "render" of the hall, then applying the generated IR on a clave sound.

I'll let people decide for themselves which sound they prefer. The clave going through the 224 concert hall, and the way it flutters at the beginning, reminds me of Vangelis doing the same thing in his 1983 classic "Antarctica", where he took the clap of a drum machine and ran it through the 224's Concert Hall with the decay at the maximum time and the same flutter at the beginning of the reverberant wash.

I could have given the sound more flutter or a slower build-up by moving the virtual microphones around and increasing the size of the virtual hall (the program can do a hall up to 75 meters on each side; this virtual hall was about 30 meters wide).

Sean: Thanks a lot for the pointers. I will download these articles and read them on my flight tomorrow. It gives me something good and worthwhile to read.

[warbler]

Very interesting!
Shouldn't the PCM96 be able to sound very close to your 224 clips?

[zmix]

Here is a simple (and certainly dated sounding) repeated drum machine figure demonstrating the oldest Lexicon and the newest Lexicon reverb's plate algo. This example also has some EQ and a little Bus Compression via the Alan Smart C2.

I tried to match the reverb settings as carefully as possible.

What I love about the 224 is how aggressive and compelling it is. It's nearly frightening!
The SONY reverbs from the 1980s have a bit of this, too.

224 parameters

PERCUSSION PLATE - A (Program button 5)

BASS: 2.8s
MID:1.4s
CROSSOVER: 200hz
TREBLE DECAY 10.9khz
DEPTH: 31
SHIFT-DEPTH (diffusion): 63
PRE DELAY: 000
MODE ENHANCEMENT: OFF
DECAY OPTIMIZATION: OFF


PCM96 Parameters

PLATE
RT 1.13
RvbOutFreq 2000hz
Diff 75%
RvbHiCut 10000hz
BassRT: 1.25x
Tailwidth: STEREO
Early Level-7.0dB
Shape: 64
Spread: 8%
RoomSize: 33.0m
BassXover 200hz
Tapslope: 0.0
Spin:0.0hz
Wander:0.0ms

L->L Reflection: 29.0ms
R->R Reflection: 23.0ms

ErlyType: TWO POLE LOWPASS
ErlyFreq: 8000.0hz
ErlyShelf: OFF

RvbOutType: BANDPASS
RvboutFreq: 2000.0hz
RvbOutBndwth: 3.0octaves
RvbOutShelf: -8.0dB

SONY :

ALGO: Plate
Rev T: 1.5s
ER 1/2 11ms
ER 3/4 31ms
Hc 0.99
Size 7
Predelay 1/2 47ms
Er level 1/2 0.992
Er level 3/4 0.695
EQ: flat
Eff Lev: 1.000

[warbler]

The 224 sounds much more interesting in that last example.
Kind of makes the 96 sound a little sterile and clinical.

Any idea why?