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.

Eno/Lanois Shimmer Sound: How it is made

The basic foundation of the Brian Eno / Daniel Lanois shimmer sound is fairly simple: Create a feedback loop, incorporating a pitch shifter set to +1 octave, and a reverb with a fairly long decay time. By controlling the gain and equalization of the feedback loop, and the lengths of the various delays within the loop, the temporal evolution of the sound can be altered from steel drum-esque sounds to the slow attack “string pads” hear on many of the Eno/Lanois tracks. This is the same technique used by ValhallaShimmer, with the reverberation, pitch shifting and feedback all incorporated within the same plugin.

Kevin Killen, answering a question about the signal flow on the U2 song “4th of July” on Gearslutz, described the signal path as follows:

The delay and modulation was derived from the AMS 1580. On its fader return , some hi frequencies were rolled off, then it was fed into a 224 Hall setting, probably 5 seconds but with a rolloff in the top and bottom. This return may have been equalised also. We may have added a second delay but then the delays have to be timed to the track as the net effect is blurring the chord progression…Our last tweak would be to play with the sends on all of the returns to the point that its almost recirculating out of control, which in turn is creating a layer upon layer effect.

The AMS DMX 15-80s was a digital delay / sampler / pitch shifter that was in common use in Britain in the early 1980′s. Eno and Lanois have both sung the praises of this unit, and Wendy Carlos has said that the AMS unit had “perhaps the least audible artifacts to pitch shifting available at that time.”

David Kulka has written that the AMS DMX had an optional de-glitch card installed, which worked on a similar principle to the auto-correlation deglitcher in the H949. His post is worth quoting:

Harmonizers, at least the early ones, had to electronically “splice” sections of the waveform in order to accomplish pitch change. When the out and in points had different voltage levels, a small DC pop could be heard at each transition. The result was a sort of low level crackle, more obvious with certain kinds of program material, and more audible at extreme pitch change settings.

The Eventide H910 exhibited this, along with the early AMS Harmonizers. Both Eventide (on the H949) and AMS partially resolved this by adding “de-glitch” cards. The circuitry on this card added a “smart” algorithm to pitch change, adjusting the transitions to better match voltages at the in and out points.

The “224 Hall setting” that Killen refers to is the Concert Hall algorithm in the Lexicon 224. This algorithm has a fairly low initial echo density, that builds to a higher density as the decay evolves. The Concert Hall algorithm is also distinguished by its high degree of modulation. The resulting sound is not a terribly accurate simulation of a real concert hall, but rather a lush and spatially expansive reverb that is still sought after more than 30 years after its introduction.

Other accounts of the “shimmer” sound refer to different reverbs being used, such as the EMT250. In addition, modulated delay lines, such as the Lexicon Prime Time, have been used by Lanois at different times. The common elements always seem to be the pitch shifter, a modulated reverb and/or a modulated delay line, and feedback and equalization generated via an analog mixer. In my next post, I will analyze the contributions of these elements to the shimmer sound, and will discuss how the various components respond in a feedback situation.

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.

Early examples of modulated reverbs

I have been trying to track down good sonic examples of a modulated reverb in action. Here’s a few that come to mind:

Vangelis, “Creation du Monde,” 1973. This is an interesting example of a home-brewed modulated reverb. Vangelis used 3 Roland Space Echos in series, processing a Hammond organ and Clavinet, to get the huge spacey sounds in this track. The old Space Echos (and other tape echos) tended to have a fair amount of wow and flutter, due to imperfections in the tape, slipping and sticking of the capstan, and so on. By applying feedback in a single tape echo, the result is a modulated echo sound, where the amount of modulation increases with each passthrough of the system. Run three of these in series, and the result is a LOT of pitch modulation.

Harold Budd and Brian Eno. “First Light,” 1980. The modulation of the EMT250 can be clearly heard on this track from “The Plateaux of Mirror.”

Jeff Buckley, “Hallelujah,” 1994. Buckley’s guitar is being run through an Alesis Quadraverb, which has several really nice modulated reverb options available (and a lot of noise, at least on my Quadraverb). The vocals sound like a modulated reverb, but I am unsure of which unit – a high end Lexicon seems likely.

I know that there are a lot more, but I am kind of forgetful today. Any suggestions welcome.

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.