Time Delays and Reverberation

In the previous module, we began investigating some of the many ways in which time delays are involved in both acoustic and electroacoustic processes, and that rather surprisingly result in a wide range of perceptual effects, ranging from timbral alterations in the very short domain (via phasing), spatial effects with reflections in the medium range, rhythmic effects with echo in a longer range, and by extension, larger patterns over time.

In the acoustic module on Sound-Environment Interaction, we provided a summary of the acoustic processes in enclosed and semi-enclosed spaces that produce the sound field known as reverberation. Multiple reflections, all of which are frequency dependent, based on the nature of the space, its walls, floor and ceiling, along with all objects within the space, combine to spread throughout the space and reinforce the sound produced within it. In a sense, reverberation is the complex aural image created by the space itself. If you are not familiar with how this process works, check out sections B and C in that module.

The main psychoacoustic effect of reverberation is to increase the perceived magnitude or volume of sounds in a space. It does so by prolonging the sound, thereby adding loudness, and spectral colouration, as well as blending multiple sounds together. In the electroacoustic world, dry synthesized sounds are often in need of such enhancement.

However, in the acoustic tradition, all soundmaking within the space must adapt itself optimally to the reverberant conditions, in terms of speed, dynamic range and timbral articulation. In other words, soundmaking, particularly with speech, cannot be independent of the acoustic space. Too much reverberation can reduce speech comprehension and muddy a musical ensemble. In subjective tests, listener preferences are aimed at combining a sense of envelopment in the right balance with definition and intimacy. Too much of one reduces the other.

In the electroacoustic world, there are no such constraints, so it is up to the sound designer’s sensibility to create the optimal balance. We will cover the topic and its applied aspects in these sub-topics.

A) Reverberation in the analog and digital domains

B) Impulse Reverberation

C) Studio demo's of reverberation

D) Studio demo's of impulse reverb

Q) Review quiz

A. Reverberation: Analog and Digital. As described in the last module, time delays, such as those involved with sound reflections, require some form of memory. In the early 20th century, such storage was difficult to implement, not only for audio but also early computing research. In the early broadcasting industry, as described by Emily Thompson in The Soundscape of Modernity, radio stations preferred to avoid using reverberation with their monophonic signals, and hence fitted out their studios with absorbent material to provide a dry acoustic. Given the problem of static and low bandwidth for the radio signal, reverberation was deemed to be a detriment for the listener – essentially a form of noise.

However, the attractive aural qualities provided by reverberation and a desire for “realism” meant that over time, various analog means of providing it were developed. Larger broadcast studios created their own echo chamber or reverb chamber on site, where the signal could be played back into the chamber and picked up by microphones again. Of course it was difficult to control the reverb time, but once mixers were available, its level could be controlled much as we do today.

Another solution that was developed in the 1940s and 50s was the spring reverberator and the plate reverberator, both of which used an electromechanical transducer to transfer the signal into a metal spring or plate at one end, and retrieve it via a contact microphone or pickup at the other end. The spring reverb unit, being smaller, was pioneered for the Hammond Organ.

Large plate reverberators, such as those produced by EMT in Germany, were more sophisticated and included a damping mechanism connected to the very large metal sheets. Smaller units were often included in electroacoustic music studios as well, but it was the use of specific units in pop music in the 1950s and 60s that established the mystique attached to the specific sound of these units. Of course, their frequency response was far from neutral, and with the smaller units, reverb time couldn’t be controlled. These specialized forms of reverb are often imitated in today’s reverb plug-ins.

Digital audio and digital reverberation started developing in the 1970s, and its techniques are beyond our scope in terms of details. One of the main pioneers, Barry Blesser, has devoted a chapter in his book, co-authored with Linda-Ruth Salter, Spaces Speak: Are You Listening, to the historical development of high-end digital reverberation and makes for interesting reading. Today, numerous reverb plug-ins are available, each with their own set of variables, ranging from simplistic to bewilderingly detailed. It is also typical that presets are offered for specific types of spaces or specific vocal or instrumental sources, such that reverb is often simply chosen from a menu, not specifically designed.

As with many issues in the electroacoustic world, justification and intention is often expressed in the language of fidelity, such as “realism”, while at the same time the operational reality is to enhance and essentially create an artificial realism. With sufficient exposure, such artificiality becomes normalized and familiar, and simply part of cultural experience. In a previous module, this was referred to as a “normalization of the artificial”. The purpose here is not to say whether this is good or bad, but simply to compare audio practice with everyday aural experience.

Here’s an interesting place to start: three examples of a mezzo-soprano voice with added reverberation. In the next section, we will discuss impulse reverberation as an example of convolution, also a form of digital processing. For each of the three examples, try to determine if it is artificially produced by a digital algorithm, or if it is modelling an actual space via convolution. In each case, we've chosen a large church acoustic with a long reverb.

(Source: Sue McGowan)

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First of all, it’s clear you have to listen carefully to hear the differences. Examples A and C are algorithmically produced, A with the ubiquitous DVerb plug-in using these settings, and C with the AirRvb plug-in using these settings. In each case the direct signal was lowered by 12 dB to simulate the distance from the mike involved in the convolved version (B) whose Impulse Response was recorded in an Italian cathedral, as seen here. Note that the AirRvb reverb time was lowered to 5 seconds, because at 8 seconds, it simply lasted too long.

Two main differences to notice are that a high-frequency boost was added to the SoundHack convolved example (B), but slightly rolled off to be less bright. The algorithmic versions, however, seem to emphasize the “bite” of the initial attack such that the voice seems closer than the convolved version which actually does sound at the distance of the microphone from the source as predicted.

In each algorithmic case, there was zero pre-delay added. This is the term used to delay the onset of the reverb to avoid masking, but it also seems to serve in this case to maintain the close presence of the voice, rather than move it back in space.

So, did you have a preference? The algorithmic versions clearly have a super smooth form of reverb, but the question remains as to whether you want this effect on everything you use it for. I think it’s clear that the impulse reverb with convolution has the advantage of sounding very different for each space that has about the same reverb time.

The frequency response of each acoustic space is highly different, but in most plug-ins, there is a limited array of spectrum controls. In the simplest cases, we get a generic choice, for instance in the following example of (1) bright (emphasized highs); (2) dark (de-emphasized highs); (3) large warm (emphasized mid-range with longer reverb time); (4) gated.

Four reverb types: bright, dark, warm, gated

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The gated case is, of course, entirely artificial as it cannot happen in the acoustic world. That is, the reverb is added only during the duration of the original sound, and removed (i.e. gated out) immediately following. This enriches the timbre of the original with no possibility of the reverb masking or muddying the following sounds.

Digital reverberation algorithms today can easily produce the kind of echo density of reflections that are required – ideally more than 1000/sec. A density that is too low produces a fluttering effect. The early digital delay lines discussed in the last module, such as the Lexicon, could only use feedback and modulation of its two delay lines, so the quality of reverberation was quite limited.

As discussed in the Sound-Environment Interaction module, good concert hall acoustics include early reflections arriving within the first 100 ms. Late arriving reflections should have smooth decay with high frequency energy falling off faster than the lows. A typical circuit proposed as early as the 1960s by Manfred Schroeder included comb filters in parallel (to simulate early reflections), and cascaded all-pass filters to synthesize reverb. However, many other models, including those involving feedback have been proposed.

In small and medium sized rooms, resonances known as eigentones are predominant because of the smaller dimensions, as discussed here. In larger rooms, reverberation is dominant. However, some tunnels exhibit both characteristics because of their length. Here is a final example of how resonance and reverberation can interact, recorded by in the vaults of the National Library in Vienna.

Tunnel inside the National Library, Vienna (Source WSP Eur 23-24)

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B. Impulse Reverberation.
Impulse reverb, also known as Convolution Reverb, is a technique that involves convolving an acoustically dry sound with the impulse response (IR) of a space. Normally the IR is a recording of a short, broadband sound with a strong attack, such as breaking a balloon, but it can also be derived from a theoretical calculation of the properties of an acoustic space based on its size and component materials.

An acoustician could use a starter pistol as a source, but for obvious reasons that is not practical or advisable for an individual. Informally, a handclap is commonly used to test out the acoustics of a space because it is short enough, and broadband enough, to hear the frequency response of the space and its reverberation time. However, for more precise testing, a standardized and repeatable sound is needed.

Reverberation time is how long it takes the sound to die away, that is, to decay to -60 dB of its original strength, but what is more important in the IR is the frequency colouration the space provides, as this can be quite complex.

You can hear a set of IR examples here in the Sound-Environment Module, if you haven’t done so already. Then return and listen to this set of examples of vocal sounds processed with them and others.

Convolution then is a mathematical model of exactly what happens to sound in an acoustic space, hence the realism of the results. It follows the principle that the spectra of the sound and the IR are multiplied together, and that the resulting duration is the sum of the durations of the sound and the IR, as you would expect from reverberation lengthening a sound. Multiplying the spectra is what we mean when we say that some frequencies are emphasized and some attenuated in an acoustically bright or dark room.

The apparent distance of the source sound that results from the convolution is the same as the distance the original source (e.g. balloon) was from the microphone that recorded it. For sound production, this can seem like a limitation, as we are used to moving sounds around in a virtual space. The options for doing this with impulse reverb (as the process is usually called) are:
- technically you need an IR recording for several positions and distances in the space; some IR catalogues provide this, but it is not common

- in the vocal sound examples in the previous webpage, there were two that showed that if you convolve a sound with the same IR twice, it will appear to be at double the distance, so that technique could be used with cross-fading

- the most common way of moving a sound closer or farther from the listener is to adjust the so-called dry/wet mix, that is, the relative proportions of the original sound and the reverberated portion. The reverberated part is usually kept constant and the dry component varied; the stronger the dry sound is, the closer the sound will seem. For very large distances, the reverb signal should also be slowly attenuated. Most impulse reverb apps will provide this, because it is easy and effective. In fact the psychoacoustic cue for distance, even in a monophonic dimension, is so strong and we are so used to it, that moving a sound towards or away from the listener is easily achieved

- in a DAW mix, you can multitrack both versions, dry and wet, that is, original and reverberated, and then adjust the level of the original in any manner desired; moreover some simple panning left and right will add lateral movement
This last suggestion is how the problem was solved that was referred to in the previous webpage about the actor and the overly long reverb time in the empty theatre where the IR was recorded. Since the dry and wet versions were easy to synch and mix, simple panning and level changes made it seem like the actor was moving around the space, but the emphasis given the dry signal level kept the text comprehensible.

Prospero's final speech, The Tempest, convolved with the Royal Drama Theatre, Stockholm
Source: Christopher Gaze

Another effect of convolution with Impulse Reverb is the smearing of attacks, as shown below. This effect occurs with reverberation because of the early reflections combining with the original sound. It is often not very noticeable, as we saw in the first sound example where we compared digital algorithmic reverb with impulse reverb. The algorithmic reverb minimized this smearing and kept the attack stronger, and seemingly closer. With auto-convolution (convolving the sound with itself) attacks are almost completely gone.

Smearing of attacks in impulse reverb

One question that often occurs to students who are learning about impulse reverb for the first time, is why we don’t hear the percussive IR in the convolved sound. The simple answer is the same as why we don’t hear reflections separately in an enclosed space – there are so many of them that they fuse together into the impression of reverberation.


C. Studio Demo of Reverberation. Reverberation in studio production is usually added within the context of a mix, particularly if plug-ins are being used. In the previous section, we raised the possibility of using Impulse Reverb as a means of processing individual sounds, presumably prior to being used in a mix, or as in the dry/wet example of Prospero’s speech, incorporating each version within a subsequent mix.

In both analog and digital mixing contexts, the Auxiliary circuit has been and still is the standard way to incorporate reverberation into a mix with each track given the option of whether it is reverberated, and if so, with what strength and characteristics. We have already encountered the Auxiliary circuit in the design of parallel processing, where a signal can be sent to multiple processors via multiple auxiliaries.

Here we use the more traditional route of using one Aux circuit to send multiple signals to the same processor, in this case a reverberator, as shown in this diagram.

In an analog mixer, each input channel has the option of sending the signal directly to an output channel AND sending it to one or more Auxiliary circuits with a level that is independent of the signal level going to the output channel. We’ll call these the mix level and the Aux send level, respectively. In other words, what this creates is a kind of submix, where all signals going to the same Auxiliary channel are mixed together with their own relative strengths, independent of what is going into the final mix itself.

The other, very important choice is the relationship between the mix level and the Aux send level, the choices always referred to as pre or post. These terms are short for pre-fader and post-fader. The distinction is:
- the “pre” setting sends the signal independent of the mix level, i.e. “before” that level, hence the use of “pre”

- the “post” setting sends the signal that is dependent on the mix level, i.e. “after” that level, hence the use of “post”
The Aux send level is going to a processor, such as a reverberator, and then it returns to the overall mix via the Aux return (which you can Solo, in order to hear it alone for fine adjustment). The Aux return level in the mix has its own fader to control how much global reverb goes into the mix. Therefore, we have two situations:
- in the “pre” setting, the processed signal always goes to the mix whether the original signal is there or not; this is useful for making the sound move into the distance, for instance, as described above in terms of the dry/wet mix

- in the “post” setting, the processed signal only goes to the mix when the original signal is there too, so fading out the original signal means fading out the reverb, in this case; this is likely to be the more usual situation
Mix demo. Here is a typical mix configuration with three stereo tracks on channels 14, 15 and 16. On each of those tracks, Aux A (or 1) has a Send activated. In the case of ProTools, only one of these is shown at a time, so this diagram has been photoshopped to include all three, just so you can see that they all have “post” selected (by not selecting “pre”). Each track has its own Send Level (which will control the signal level of each track being sent to the reverberator), and each Aux Send is going to a particular output “bus” (which is basically a virtual patch cord that connects the signal to the Aux channel shown to the right of the signal channels.

This Aux channel (highlighted in the bottom right corner), which receives its signal from the same bus, has an insert selected, which is the stereo DVerb plug-in. Its output goes into the overall mix (channels 1&2). Check that you understand the routing involved by enlarging the diagram and using the zoom tool if necessary.

Note that if the DAW software (stupidly) labels the output as going to a specific processor (in this case, a compressor), you can ignore this and add the processor of your choice as an Insert. Don't let yourself be "dumbed down"!

Three source mix with reverb (Click to enlarge)

This demo mix uses three soundfiles we have generated in previous exercises, ones that actually don’t make much logical sense in combination: (1) the high-pass scything sound; (2) the feedback circuit mix that combines rhythmic hammering with the percussive PVC pipe; (3) the mix of waves used in the parallel circuit. However, we accept the challenge of trying to make “aural” sense of these three semantically unrelated sounds.

Moreover, why would anyone want to put reverb on the scythe and the waves? They clearly are not going to be recognized as belonging to the same acoustic space! But they both do have rhythmic noisy timbres, so we can play on that.

There are three versions of our mix: (1) no reverb, so the illogical elements stay quite separate; (2) a mix with 4.5 seconds of reverb, but note that each Aux send has a different level, more going to the scythe, medium going to the rhythmic mix, and less going to the waves; (3) we raise the reverb level by a factor of more than 2 to about 10 seconds. You may also notice that a bit of care has been taken in placing the rhythmic repetitions of the scythe against the rhythms of the feedback circuit and the waves.

Mix with no reverb

Mix with medium reverb

Mix with high reverb

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In these mixes, no attempt was made to adjust the mix levels (at least not yet, stay tuned), except for a simple fade-in and out at each end. How did you find the balance of the elements? Which version did you prefer? Despite the illogical nature of the mix from a semantic perspective, I think mix 2 is the best, because the reverb, however, incongruent, helps to blend the three tracks into a unity, supports the build-up of rhythmic energy towards the end, and still allows individual tracks to be heard clearly.

Mix 3 is “swimming” in reverb, and possibly drowning the component sounds. Think of the balance between envelopment and definition referred to earlier – this mix skews the balance towards envelopment at the expense of clarity.

Auxiliary send in Pre mode. Here is a simple example of how to make the sound appear to recede in the distance by simply fading out the unmodified signal, while keeping a constant level of reverb. As described above, the “pre” setting allows this to be easily accomplished. In the example the mix level has been latched to fade out (doing this by ear is a good idea, instead of using breakpoints).

The single Aux circuit is in pre mode, and the send level is at 0 dB. The Dverb is set to “large church”, similar to the convolved version heard earlier, with a 8.7 sec reverb time. It would be a bit more accurate in terms of how this works in the real world, to slowly attenuate the reverb level as well, but much more slowly, on the Aux return (which can also be latched).

The principle here is that in an acoustic space, the direct signal from the source falls off at 6 dB per doubling of distance (the inverse-square law), but the reverberated portion falls off much more slowly so that the ratio of dry/wet gradually favours a larger percentage of the sound being the reverberated portion.

Click to enlarge
Voice appearing to recede


D. Studio demo of Impulse Reverb. Most app’s that offer Impulse Reverb or Convolution Reverb have a simple set of parameters, including the wet/dry mix which avoids having to use multiple IR files recorded at different distances. Also, look for a pre-delay option, and the type of EQ or filtering that is offered. Some app’s provide an extensive catalogue of Impulse Responses, and the option of importing others. However, if the software doesn't allow the reverb process to be extended to auto-convolution, or any other arbitrary combination of files, it is going to be limited in terms of what we are developing here.

However, keep in mind, that you may have to add several seconds of silence to your file if you don’t want to have it cut off once you apply it to your sound – a typical demonstration of how inflexible the processing paradigm is, despite the obvious need to incorporate this into your processing. This problem will likely be true of DAW’s as well as editors.

"ImpulseVerb" in the Peak editor

Some editors may also allow you to use the “impulse” from the clipboard. This means you load the IR (or the file itself for auto-convolution) into the editor first, copy it to the clipboard (command C), and then use it for the Impulse Reverb. Since auto-convolution doubles the length of the output file, you will need to add that much silence to the soundfile, all of which makes one appreciate SoundHack more, since its author, Tom Erbe, obviously knows what is really involved with convolution.

With SoundHack, Impulse Reverb is a typical process of loading the source file (command O), selecting Convolution from the Hack menu (command C), at which point your file will be labelled “Input”, then you can add Normalization and (optionally) Brighten, and select the Impulse File, which is then labelled as “Impulse”.

SoundHack Convolution

At this point you’re ready to process the file, giving it a more compact name than the suggested one, and choosing 24 bit output. You can test it out by playing it in the bar at the bottom, but if you want to edit it, delete the file from SoundHack on the desktop, and open the file in your editor (i.e. you can't have two versions of the output file open at the same time). As noted above, if you want to further move your sound around in the space you've created, multi-track the original with this processed version and adjust levels and panning.

Note: SoundHack convolution will only work with .aif files (not .wav) on the Mac.


Q. Try this review quiz to test your comprehension of the above material, and perhaps to clarify some distinctions you may have missed.