Slides from my AES Reverb presentation

I was invited by the Seattle chapter of the Audio Engineering Society to speak about reverbs and reverb design. I threw together some slides:


Pretty skeletal deck, but it was (hopefully) more entertaining when presented in person. Don Gunn helped me out with a Logic X project that accompanied the presentation. Don also listened to me when I practiced the presentation, and was graceful enough to pretend that he hasn’t heard me ranting on the same topics about 100 times before.

I think I learned more from the people in the audience at the AES presentation than anyone learned from me! It was cool to hear anecdotes from people that had worked at Lexicon and Alesis, as well as folks that had a lot of experience with plate reverbs and echo chambers. Thanks to Christopher Deckard for inviting me to speak, and thanks to everyone that attended the presentation.

Reverbs: Diffusion, allpass delays, and metallic artifacts

One of the most common controls found in reverberation algorithms is the Diffusion control. This is usually described as increasing the echo density, either the initial echo density (for Lexicon algorithms) or the rate at which echo density builds over time. The manual for the Lexicon LXP-15 has a somewhat typical description of the Diffusion parameter:

Diffusion: Controls the degree to which initial echo density increases over time. High settings of Diffusion result in high initial buildup of echo density; low settings cause low initial buildup. Echo density is affected by Size — smaller spaces will sound denser. To enhance percussion, use high settings of Diffusion. For clearer and more natural vocals, mixes and piano music, use low or moderate Diffusion settings.

If you read a lot of manuals for reverb products, you will often see similar descriptions of the Diffusion control, as well as the recommendation to use lower settings of Diffusion for clearer vocals. But why is this? A real room or hall tends to start with very high levels of diffusion, due to the objects typically found in the space – chairs, furniture, intricate wall patterns, etc. It would seem that a given echo density should be a characteristic of the space, not of the signal being sent into that space.

The answer lies in the signal processing tricks used to generate the initial high echo density. Manfred Schroeder, in his seminal 1962 AES paper “Natural Sounding Artificial Reverberation,” discusses using very short feedback delay lines in series to increase the echo density. Schroeder developed a very clever feedback/feedforward trick, such that the resulting delay line has a “flat” frequency response. The resulting delay unit is referred to as an allpass delay:

In the late 1970’s, James Moorer published an optimized version of the allpass delay, which used less multiplies, and is more commonly used today:

The earliest commercial digital reverbs, such as the EMT-250 and Lexicon 224, made use of several series allpasses at the inputs of the reverberation algorithms to increase the echo density. Lexicon was the first company to allow the user to directly control the coefficients of the input allpasses, and labeled this the “Diffusion” control.  The practice quickly spread through the audio industry.

EDIT: Chuck Zwicky, in a comment to this post, points out that the Diffusion parameter wasn’t originally present in the Lexicon 224, but was introduced with the Version 4.0 software. He also points out that most of the successful early reverberators up to 1984 did not have adjustable diffusion. The Eventide SP2016 had adjustable diffusion for some of their reverb algorithms, but this would have been around the 1984 to 1985 time frame.

The problem with generating echo density through series allpass delays stems from the definition of “allpass.” An allpass system will pass all frequencies with equal amplitude, over time. There is no guarantee when a given frequency will make its way out of the allpass delay. In practice, allpass delays don’t sound flat. Much like comb filters, a short impulsive sound sent through an allpass delay will result in a “ringing” sound, where only certain frequencies are resonating. Run an impulsive signal through several short allpass delays in series, and the result is a metallic decaying sound.

For percussive instruments, the metallic coloration might be an acceptable tradeoff, versus the “chattering” sound that occurs when the initial echo density is too low. Plus, snare drums have a metallic coloration in their own right, so a bit more coloration is OK. For vocals, the coloration produced by short allpass delays can be very unpleasant. Even though vocals are usually perceived as a “smooth” or continuous signal, the actual waveform produced by the glottis is very pulse-like, and can cause short series allpasses to ring out. This is especially audible on male vocals.

Some of the possible solutions to the issues with series allpasses:

  • Embrace the metallic coloration, use a bunch of series allpasses, and call the resulting algorithm a plate reverb. This is a fairly common approach, with most of the “plate” algorithms having very little to do with a physical plate, so much as having a lot of initial echo density and a somewhat metallic sound.
  • Use fewer series allpasses at the input. This works in eliminating coloration, but can result in a lower initial echo density. Many “hall” algorithms use this trick.
  • Use a larger number of series allpasses, with the idea being that the larger number of resonances will end up smearing out the metallic sound. This works, but a side effect of cascading a larger number of series allpasses is that the attack time can be extended to the point where the sound seems to “fade in.” This is a great sound if you like it, but doesn’t work for small room simulation.
  • Modulate the delay lengths within the allpasses. For longer allpasses, this helps reduce coloration. For the short allpasses used in the input diffusion section, this ends up producing too audible of a chorusing sound, or a sound similar to water sloshing around in a metal pan.
  • Reduce the coefficients of the allpass delays. This will reduce coloration, but will also reduce the echo density.

This is where the Diffusion control comes in. Instead of being a compromise solution that works OK for all signals and not great for any signal, it allows the user to adjust the algorithm to suit the input signal. It places the burden of balancing echo density and coloration on the end user, instead of on the algorithm designer. By knowing how the Diffusion control works, the end user can make their reverbs work better for them.

Is this an ideal solution? Probably not. But in the limited hardware processors of the late 1970’s, or the low-CPU plugins of today, it can be a reasonably effective solution.

EDIT #2: ValhallaRoom uses some clever signal processing tricks to avoid the issues associated with series allpass delays described above. A high level explanation of the Early Reverb section of ValhallaRoom can be found here. Even though ValhallaRoom has a Diffusion control, it is not being used to control allpass coefficients – the Early Reverb has no allpasses in it.

EDIT#3: ValhallaShimmer is built around a large number of cascaded, modulated allpass delays, and the artifacts that are generated by such a structure (see this blog post for more details). In addition, many of the “classic” digital reverbs relied heavily on series allpasses, so it isn’t to say that they produce a sound that is unusable – just that this sound isn’t necessarily reflective of what is found in a “real” acoustic space.

The Reverb Beard

Something that I find rather curious, is that many of the reverb pioneers sported some seriously impressive beards. Christopher Moore has posted a few beard-heavy pictures on his website ( Here’s my favorite:

From left to right, you have Christopher Moore (Ursa Major reverbs, AKG ADR 68K), Anthony Agnello (Eventide, Princeton Digital), Wolfgang Schwarz (or Wolfgang Buchleitner, not sure of the name, but the Quantec guy) and David Grieisinger (designer of the Lexicon reverb algorithms). An amazing amount of reverb knowledge in one place, and rocking beards that rival ZZ Top, assuming that you put the 4 beards together to form one super beard like some sort of beard Voltron.

Another picture tosses in Barry Blesser (EMT-250), sporting a scholarly pipe and an even more scholarly beard:

Nowadays, I use the term “Reverb Beard” (or “Reverbskägg” in Swedish) to refer to people that develop reverberation algorithms, or to describe the state of people in the middle of the design process for reverb algorithms. Feel free to use this meme.

Note: I’ve tried to grow the reverb beard before, but it either comes out red, which makes me look like Kris Kringle in “Santa Claus is Coming To Town,” or greyish-red, which makes me look and feel old. So the “reverb beard” is more of a mental state.

EDIT: Chris Randall and Adam Schabtach, of Audio Damage fame, both pointed out to me that the mighty beards of the reverb pioneers were first mentioned on the Music Thing blog in 2004:

The first digital pitch shifter: Lexicon Varispeech

When I was planning my “editorial calendar” for the next few weeks, I had planned on discussing the Eventide H910 Harmonizer as the “first digital pitch shifter.” I even described the H910 as such in an earlier blog post. However, it turns out I was wrong. The Lee article that I discussed in my previous post describes what is probably the first commercially available pitch shifter, the Lexicon Varispeech:

The Lexicon Varispeech was introduced in 1972, a good 3 years before the H910.  The Lexicon Pro website makes only passing mention of the device, describing it as a “Lexicon product for the language instruction market.” Fortunately, the Obsoletetechnology blog has a nice overview of the device, including photos, gutshots, and sound examples. The following image is taken directly from the aforementioned blog post, which you really should read:

Interestingly enough, for a device that was marketed as being used for speech and time compression, the Varispeech 27Y has a feedback knob. This is solely for use as a special effect, and was prominently featured on the H910 and H949 harmonizers of later years. I am uncertain if this was in the 1972 Varispeech, or if the 27Y was the original Lexicon model or a later version. If anyone has any info, please contact me.

Chris Walla of Death Cab for Cutie describes his use of the Lexicon Varispeech in an EQ Mag interview, where he also notes the incongruity of the feedback knob on a device used for time compression and expansion:

There was a lot of speech pathology research developed at Lexicon that was cross-purposed into pro audio. The Varispeech was originally intended to help stroke victims and people with speech disorders. The idea was that you could slow down a conversation at regular pitch but keep pitch where it was so that people could practice figuring out how to reconnect their mouth and their brain.

There was this weird period where [Lexicon was] screwing around with it; I got one that had a feedback knob, which as far as I can tell is completely useless for speech pathology, but it makes everything sound like Doctor Who, which is awesome.

It sounds great under the snare drum, and Tegan’s vocals run through it on ‘The Cure’ when she does the ‘Oh, uh oh, uh oh’ thing. The Varispeech is a really cool chorus-y, flange-y thing if you set it up that way. But it’s a speaker destroyer, too. It’s an old [’70s] effect, and Lexicon wasn’t worried about being sued by guys who were like, ‘You blew up my guitar amp, dude!’

Eos tips and tricks: Recreating the PCM70 Tiled Room

One of the “greatest hits” of artificial reverbs is the Tiled Room preset in the V2 ROM of the Lexicon PCM70. This preset went missing from the V3 software, but you can find a listing of the parameters here.

I was playing with an impulse response of this preset, and decided to emulate it in Eos. Here’s what I came up with:

  • Type: Plate 2. For maximum authenticity, you could use Plate 1, as both Plate 1 and the Rich Chamber algorithm used in the PCM70 Tiled Room preset have a mono input. However, Plate 2 works better with stereo miked material.
  • Pre-Delay Time: 4 msec. This is the same as the original Tiled Room preset. If you wanted to simulate a somewhat bigger space (or the delayed room mikes used by many engineers, such as Steve Albini for The Jesus Lizard’s “Goat”), you could set this between 15 and 30 msec.
  • Size: Try setting this between 10 and 20 meters. The PCM70 preset used 8 meters, but this might have been an attempt to get a more exponential decay out of the Rich Chamber algorithm, which has a fairly “flat” decay compared to most acoustic spaces. The Plate algorithms in Eos will always decay away exponentially, so Size can be used to control the apparent size of the room, as well as the desired coloration.
  • Attack: Set this fairly low.
  • Diffusion: I used 0.5. For drums, you may wish to increase Diffusion, while vocals might require lower settings to avoid metallic coloration. This is true with the Lexicon as well, and seems to have something to do with the pulsetrain waveform of vocals. The Plate algorithms in Eos have a fairly high initial echo density to begin with, even with lower settings of Diffusion.
  • Decay: 0.62. Same as the Lexicon.
  • Low Multipler: 1.258. This results in the same running decay time as the Lexicon
  • Low Crossover (accessible in the automation view in some programs, and the Controls View in Logic): 3410 Hz.
  • High Multipler: 0.25. The Lexicon Rich Chamber algorithm uses a one-pole lowpass filter, as opposed to the 1st order shelf in Eos, so setting the High Multiplier to 0.25 better emulates the steeper rolloff of the Lexicon
  • High Crossover: 9000. This is pretty high, but it emulates the setting in the Tiled Room preset.
  • High Cut: 10000 Hz. I would probably set this lower as needed. The PCM70 has a hard cutoff of 15 kHz, so you may wish to lower the High Cut setting to compensate for this.

The other settings have no corresponding settings in the PCM70, so adjust for taste:

  • Low Cut should be used to eliminate any unwanted “booming” of the bass frequencies.
  • Mod Rate and Mod Depth aren’t relevant to emulating the Tiled Room preset, as the Rich Chamber algorithm in the PCM70 didn’t have modulation – but if you like the sound of modulation, go for it. Smaller Size settings will result in more apparent modulation.
  • The stopped reverb decay will be shorter on the PCM70 than in Eos (or in the PCM70 impulse responses), as the PCM70 has separate decay settings for stopped reverb, so adjust the Eos decay time as needed to strike a good balance between running decay and stopped decay.

More general reverb tips

As a followup to my Eos tips and tricks post, I thought I’d share some more reverb tips. All of these have been tested with Eos, but should also work with a wide variety of hardware and software reverbs.

  • Set the high cut filters to a fairly low frequency. Older hardware reverbs, such as the EMT250 and Lexicon 224, had a hard cutoff at 8 KHz to 10 KHz, due to the low sampling rate of the machines. The high cut filters in many reverbs have a much more gentle slope than the high order filters used at the inputs of these old boxes. To emulate these old boxes, try setting your high cut filters to a fairly low frequency, such as 2 to 4 KHz. This also corresponds more closely to the absorbtion of high frequencies by air in a large space, such as a concert hall.
  • Use the low cut controls to make the reverb sound less “tubby.” Many concert halls actually have a much longer decay for low frequencies than mid range frequencies. This is useful for classical music, but for most popular music forms, the amount of bass energy that is present will sound flubby when reverberated. Set the low cut controls at 200 to 400 Hz, or even higher, for a clearer reverb sound.
  • If you don’t have low or high cut controls, put the reverb in a send bus, and put the EQ of your choice before or after the reverb.
  • Try compressing the input or output of the reverb, for some neat sounds. A limiter before the reverb will keep spikey transients from dominating the response, and will better emulate the transformer-coupled inputs of the old high-end hardware units. Compressing the output of the reverb will change the exponential decay response to something much weirder, depending on your settings.
  • Plate reverbs have a lot of high frequencies in the decay, so make sure that the high frequency decay multiplier (or the high frequency decay filter cutoff) is set fairly high. These controls are usually separate from the high cut controls, that shape the sound at the input (or output) of the reverb.
  • Adjust your modulation depth based on the decay. For long decays, you may wish to back off on the modulation depth, as the sound will travel through the modulators many more times compared to a short decay. Each pass through the modulators causes more detuning. A modulation depth that works for short decays may sound seasick for long decays. Of course, if that’s your thing, then go for it.
  • Use the Size control for the desired echo density, but be mindful of how it affects the modal density. For example, if you want a small drum room, then set Size to a smaller setting, as it will make the echos closer together. However, a smaller Size setting will sound more metallic for longer decays, as the modal density goes down as the Size decreases. Longer delay lengths = higher modal density = less metallic = lower echo density. For short decays, the low modal density may not matter.
  • The Size control is often given in meters. This has nothing to do with any real physical world metrics, in most cases. A real acoustic space with a 30 meter maximum dimension will have a few orders of magnitude higher modal density than your typical digital reverb with a 30 meter Size setting. Just tune it by ear to where you like it.
  • Shorter Size settings may also result in deeper modulation for the same decay setting, so be sure to retune this for your tastes.

Hope these are useful to people. If you have any more tips, feel free to add them in the comments.

Modulation in reverbs: reality and unreality

The use of modulation in digital reverbs dates back to the first commercial digital reverberators. The EMT250 used an enormous amount of modulation, to the point where it sounded like a chorus unit. Lexicon’s 224 reverberator incorporated what they called “chorus” into the algorithms, working along principles not dissimilar to the string ensembles in use at the time. The Ursa Major Space Station was based around an unstable feedback arrangement, that relied upon randomization to achieve longer decay times without self-oscillating.

Recently, Barry Blesser has written about randomization in his book, “Spaces Speak: Are You Listening?” Blesser argues that thermal variations in most real-world acoustic spaces results in small variations of the speed of sound within those spaces. Multiply this by several orders of reflections, and the result is an acoustic space that is naturally time varying. Blesser goes on to argue that random time variation in algorithmic reverbs emulates the realities of an acoustic space more accurately than time-invariant convolution reverbs.

Blesser makes a convincing argument, but I am not convinced that the heavy amounts of delay modulation used in the older reverbs makes for a more “realistic” space. The randomization in the older algorithms does a nice job in masking the periodic artifacts that can be found when using a small amount of delay memory. However, the depth of modulation used in the old units goes far beyond what can be heard in any “real world” acoustic space. The thermal currents in a symphony hall will result in a slight spread of frequencies as the sound decays, but will not create the extreme chorusing and detuning found in the EMT250, or in the Lexicon algorithms with high levels of Chorus.

Having said that, I would argue that the strengths of algorithmic reverbs is not in emulating “real” acoustic spaces, but in creating new acoustic spaces that never existed before. Blesser recently said that the marketing angle of the EMT250 was to reproduce the sound of a concert hall, but later describes the EMT250 in terms of a “pure effect world.” The early digital reverbs, in the hands of sonic innovators such as Brian Eno and Daniel Lanois, were quickly put towards the goal of generating an unreal ambience, where sounds hang in space, slowly evolving and modulating. Listen to Brian Eno’s work with Harold Budd, on “The Plateaux of Mirror,” to hear the long ambiences and heavy chorusing of the EMT250 in action. A later generation of ambient artists made heavy use of the modulated reverb algorithms in such boxes as the Alesis Quadraverb to create sheets of sound, that bear little resemblance to any acoustic space found on earth.

Creating these washy, chorused, “spacey” reverbs has been a pursuit of mine since 1999. My early Csound work explored relatively simple feedback delay networks, with randomly modulated delay lengths, in order to achieve huge reverb decays that turn any input signal into “spectral plasma” (a term lifted from Christopher Moore, the Ursa Major reverb designer). With my more recent work, I have tried to strike a balance between realistic reverberation, and the unrealistic sounds of the early digital units. The plate algorithms in Eos are an attempt to emulate the natural exponential decay of a metal plate, but were also inspired by my understanding of the EMT250. The Superhall algorithm in Eos was not attempting to emulate any “natural” space, but rather the classic early digital hall algorithms, with heavy randomization, nonlinear build of the initial reverberation decay, and the possibility of obtaining near infinite decays. The “real” world continues to be a source of inspriation for my algorithms, but I find myself more attracted to the unreal side.