I learned so very much starting out as a producer that I really want to give back to you guys trying to figure things out for the first time:

DISCLAIMER

THESE NOTES ARE NOT MY OWN ORIGINAL NOTES

These are a bunch of notes that I have consolidated from a great text in audio mixing techniques, written by Roey Izhaki
. You can find the full text (and I very much suggest that you do, it's CHOCK full of great information) HERE

.

anyway, without much further ado...

REVERB

Reverb can be used to resolve masking. The common usage of reverbs for the purpose of depth gives us the ability to position sounds in a front–back fashion in addition to the conventional left–right panning. This gives another dimension and much more space in which we can place our instruments. Sometimes reverbs are added just to spice things up and add some interest. Snare reverb, reversed reverb, gated reverb or many other reverbs are simply added based on our creative vision and not necessarily based on a practical mixing need. Reverbs can be automated between sections of the production in order to create some movement or be introduced occasionally to mark transitions between different sections.

Spring Reverb

Quiet operation and flat frequency response were never an asset of true electromechanical spring reverbs, and they are known to produce an unpleasant sound when fed with transients. Therefore, spring reverbs are usually applied on material like pads and vocals, normally while only mixing a small amount of the wet signal.

Plate reverb

If a spring reverb can be considered as a ’one-dimensional’ reverb simulator, the logical enhancement to such a primitive design was going two-dimensional. The operation of a plate reverb is very similar to that of a spring reverb, only that the actual vibrations are transmitted over a thin metal plate suspended in a wooden box. An input transducer excites the plate and output transducers are used to pick up the vibrations. In the case of plates, these vibration propagate on its two-dimensional surface and bounce from its edges. Whilst mobility was not one of its main features, it had better sonic qualities than the spring reverb. In addition to a damping mechanism that enabled control over the decay time, its frequency response was more musical. The reverb it produced, although still not resembling a true natural reverb and being slightly metallic, blended well with virtually every instrument, especially with vocals. The bright, dense and smooth character of plate reverbs also made them a likeable choice for drums, which explains why they are so frequently added to snares.

Digital emulators

A digital reverb is not a dense collection of simple delays, since these will cause much coloration to the direct sound and produce a broken frequency response. There are many internal designs for digital reverbs and they vary in their building blocks and in the way these blocks are connected. At the lowest level, digital emulators are implemented using mathematical functions that are executed on internal DSP chips. To differ them from digital convolution reverbs (soon discussed), these type of reverbs are now referred to as algorithmic reverbs. No digital emulator will ever be able to produce a reverb completely identical to the one created in a real room. This is much due to the complexity of such a reverb – there are thousands of reflections to account for, different frequencies propagate and diffract in a different fashion, different materials diffuse and absorb sound in a different way, even the air itself changes the sound as it travels through it. Manufactures must take shortcuts, and the more processing power in their disposal the less shortcuts they have to take, thus the more realistic the reverb is likely to be. As algorithmic reverbs have no physical or mechanical limitations, they provide a multitude of controls that let us tweak nearly every property of the reverb. This makes them an extremely flexible and versatile tool in many mixing scenarios, and it is no wonder that they are the most common reverbs in our mixing arsenal. An issue with high-quality emulators is that they are expensive both in terms of price and processing-power consumption.

Convolution (sampling) reverbs

Even during the 1970 s various people have toyed with the possibility of capturing the reverb characteristics of an acoustic space, so it can be applied later to any kind of recording. Reverb sampling is normally done by placing a stereo microphone pair in a room, then recording the room response (i.e., the reverb) to a very short impulse like that of a starter pistol. Since it is very hard to generate a perfect impulse, an alternative method involves playing a sine sweep through speakers instead. The original sound might be removed from the recording, leaving the reverb only. The recorded impulse response (IR) is then loaded into a convolution reverb, which analyzes it and constructs a huge mathematical matrix that is later used by the convolution formula. With every sound fed through the unit, a reverb very similar to that of the original space is produced. An emulator can be based on one of two types of convolution, either one that is done in the time domain (pure convolution) or one that is based on the frequency domain (convolution or Fourier based) – each generates the same result, only in some situations one will be faster than the other. If pure convolution is used, an impulse response of 6 seconds at 44.1kHz would require around 23 billion mathematical operations per second – an equivalent to the processing power offered by a 2.2 GHz processor. It can be easily seen how such a process might be unwieldy in some situations. As a general rule, the shorter the original impulse response is, the less the processing needed would be. Many impulse responses can also be downloaded from the Internet, many are free. The quality of the impulse recording is determined by the quality of the equipment used, which is a vital factor if natural results are sought after. It is generally agreed that a good impulse recording produces an extremely believable reverb simulation that matches (if not exceeds) the quality of the best algorithmic emulators.

Category:

Halls - Natural - Large, natural sounding, live spaces

Chambers - Natural - Simulate reverb chambers or spaces that have slightly less natural reverb behavior and less defined size

Church/Cathedral - Natural - These types of spaces might produce a highly impressive reverb for certain types of materials like organs, but they generally result in poor intelligibility

Rooms - Natural - Normal rooms of different sizes

Ambiance - Natural - Concerned more with placing the sound naturally in a virtual space, caring less about the actual reverberation. Most often an ambiance preset involves early reflections only

Plate - Effect - Plate Reverb

The choice of category is usually determined by two main factors. First, the size of the room should co-exist with the type of music. A classical recording or a chill-out production might use a large space such as a hall; a Bossanova or jazz production can benefit from a moderate-size hall, while a heavy metal or trance productions might make use of very small space simulations. Second, the decay time (or length) of the reverb should be relative to the mix density – a dense mix will suffer from long reverb tails that will cause masking and possibly clutter; in a sparse mix, longer decays can fill empty spaces. A reverb that sounds good when soloed might not interact well with the rest of the mix – final reverb selection is better done in mix context. When a reverb is used as an effect, an artistic judgment is required since there are no golden rules. Many use plate programs on vocals and drums; snares are commonly treated with a plate reverb or a gated one. For more impressive effect, chambers or halls can work. Halls can also be suitable for orchestral instruments such as strings, flutes, brass and the less orchestral instrument, the saxophone. Bass guitars are usually kept dry, and distorted electric guitars can benefit from a subtle amount of a small room reverb that will only add a touch of shine and space. As the association between synthesized sounds and natural acoustic spaces is loose, chamber and ambiance programs might be more suitable in a sequenced production. The above should merely serve as guidelines – the choice of a reverb program is truly subject to experiment and varies for each individual mix.

Pre-delay

Pre-delay gives us a certain clue regarding the size of the room, where in larger rooms the pre-delay is longer as it takes more time for reflections to travel to the boundaries and back to the listener. Pre-delay also gives us a certain clue regarding the distance between the source and the listener, but here an opposite effect takes place to what might initially seem: The closer the source to the listener the longer the pre-delay. This is due to the fact that the relative distance between the direct and the reflected sounds is getting smaller the further away the source is from the listener. This phenomenon is commonly put into practice when a reverb is required but not the depth that comes with it – we simply lengthen the pre-delay. Pre-delay is normally expressed in milliseconds, and for natural results our brain requires that it is kept below 50ms. However, longer pre-delay times are still used, for example, when trying to keep things in the front of the mix. The pre-delay time also determines when reflections start to mix with the direct sound. Reflections caused by a real room are far more complex than those produced by an emulator with its limited building blocks. The reflections generated by a digital emulator might cause comb-filtering and other side effects when mixed with the original signal. The sooner they are mixed, the more profound this effect will be. It is worth remembering that even the sound of a snare hit can easily last for 80 ms before entering its decay stage. If the early reflections are mixed with the direct sound within the very first milliseconds (like they mostly do in nature) the timbre might distort. We lengthen the pre-delay time in order to minimize this effect. On the same basis, early reflections can be masked by the original signal. Since the early reflections give us most of the information regarding the properties of the space, lengthening the pre-delay can nudge these reflections outside the masking zone and result in clearer perception of important psychoacoustic cues. Another issue related to short pre-delay settings is intelligibility – if the reverb is mixed with direct sound very early, it might blur and harm both clarity and definition. This is extremely relevant when adding reverb to vocals. A very long pre-delay can cause an audible gap between the original and the reverb sounds. This will separate the reverb from the dry sound, cause the reverb to appear behind the dry material and usually produce an unnatural effect along with possible rhythmical disorder. We usually aim at a pre-delay time that will produce minimal timbre distortion without audible time gaps. Nevertheless, audible pre-delay gaps have been used before in order to achieve interesting effects, including rhythmical ones.

Early reflections (ER)

Shortly after the direct sound, bounced reflections start arriving to the listener. Most of the early reflections only bounce from one or two surfaces, and they arrive at relatively long time intervals (a few milliseconds). Therefore, our brain identifies them as discrete sounds that are correlated to the original signal. The early reflections are indispensable in providing our brain information regarding the space characteristics and the distance between the source and the listener. A faithful early reflection pattern is obligatory if a natural simulation is required, and alterations to this parameter can greatly enhance or damage the realism of the reverb. Neither spring nor plate reverbs have distinct early reflections due to their small size, a fact for which they do not excel at delivering spatial information. With dependence on different room properties, early reflections might arrive within the first 100 ms following the direct sound. It is worth remembering that early reflections within the first 35 ms fall into the Haas zone and hence our brain discerns them in a slightly different manner. In addition, these very early reflections are readily masked by the direct sound. As discussed earlier, we can increase the pre-delay time in order to make the early reflections clearer. The level of the early reflections suggests how big the room is – a bigger room will have its boundaries further away from the listener, thus bounced reflections will travel longer distances and will be quieter. Surface materials also affect the level of reflections. For example, reflections from a concrete floor would be louder than reflections from a carpet. With relation to depth, the level of early reflections might again have an opposite effect to what initially seems. Although the further away the listener is from the sound source the longer distance the reflected sound travels, it is the difference in travel distance between the direct and the reflected sounds that matters here – a close sound source will have a very short direct path but a long reflected path. The further away the source is from the listener the smaller will be the difference in distance between the two paths. In practice, the further away the source and the listener are, the closer in level will be the direct and reflected sounds or in relative phrasing: the louder will be the early reflections. Early reflections are the closest sound to the dry sound, hence they are the main cause for timbre distortion and comb-filtering. Sometimes attenuating or removing the early reflections altogether can yield better results and more healthy timbre. Finally, one trick involves adding the early reflections alone to a dry signal. This can enliven dry recordings in a very transparent way and with little side effects. The explanation for this phenomenon goes back to the use of delays to open up sounds and create some spaciousness.

Reverberation (late reflections)

The reason that the term reverberation is used in this text and not late reflections is that the term ’reflection’ suggests something distinct like an echo. However, the reflection pattern succeeding the early reflections is so dense that it can be regarded as one bulk of sound. Sometimes, reverberation is referred to as the reverb tail. As sound is absorbed every time it encounters a surface, later reflections are absorbed more as they encounter a growing amount of surfaces. This results in reverberation with decaying amplitude. The level of the reverberation is an important factor in our perception of depth and will be explained shortly. The inverse square law defines how sound drops in amplitude with relation to the distance it travels. For example, if the sound 1m away from a speaker is 60dBSPL, the sound 5 m away from the speaker will drop by 14 dB (to 46 dBSPL), and the sound 10 m away from a speaker will drop by 20 dB (to 40 dBSPL). It should be clear that the further away the listener is from the sound source the lower in level will be the direct sound. If a listener in such a room stands 1 m away from the speaker he/she will hear the direct sound at 60dBSPL and the reverberation at 43dBSPL. The further away the listener gets from the speaker the quieter becomes the direct sound, but the reverberation level remains the same. Put another way, the further away the listener is the lower the ratio is between the direct sound and the reverberation. At 5 m away from the speaker the listener will hear the direct sound and the reverberation at equal level. We call the distance at which such equality happens critical distance. The direct-to-reverberant ratio is commonly used by recording engineers, especially orchestral ones, to determine the perceived depth of the sound source captured by a stereo microphone pair. For mixing purposes, we decrease the level ratio between the dry signal and the reverberation in order to place instruments further away in the mix. As many emulators do not have a separate level control for reverberation, we more often compromise and adjust the ratio between the dry signal and the wet signal (which contains both early reflections and reverberation). Just like in nature, intelligibility and definition can suffer if the reverb is louder than that of the dry signal. It should be added that the perceived loudness of the reverb is also dependent on its decay time and its density.

Decay time

How long does it take a reverb to disappear? In acoustics, a measurement called RT60 is used, which states the time it takes sound in a room to decay by 60 dB. In practical terms, 60dB is the difference between a very loud sound to one that is barely audible. This measurement is also used in reverb emulators to determine the ’length’ of the reverb. Scientifically speaking, the decay time should be measured in relation to the direct sound, but some emulators reference it to the level of the first early reflection. In nature, a small absorbent room will have a decay time of around 200 ms, while a very large arena can have a decay time of approximately 4 seconds. Decay time gives us a hint regarding the size of a room – bigger rooms have longer decay times as the distance between surfaces is bigger and it takes more time for the reflections to diminish. Decay time also gives us an idea of how reflective the materials in the room are – painted tiles reflect more sound energy compared to glass wool. The decay time in a digital emulator is largely determined by the size of the room. One should expect a longer decay time in a church program than in a small room. Longer decay creates a heavier ambiance, while a shorter decay is associated with tightness. Longer decay also means a louder reverb that will cause more masking and possibly intelligibility issues. With vocals, there is a chance that the reverb of one word will overlap with the next word. This is more profound in busy-fast mixes where there are little time gaps between sounds. If too much reverb is applied in a mix and if decays are too long, we say that the mix is ’washed’ with reverb. In many cases, especially when reverbs are used on percussive instruments, the decay time should make rhythmical sense. For example, it might make sense to have a snare reverb dying out before the next kick. This will ’clear the stage’ for the kick and also reduce possible masking. There is very little point delving into time calculations here – the ear is a better musical judge than any calculator, especially when it comes to rhythmical feel. In addition, reverbs may become inaudible long before they truly diminish. Snare reverbs are commonly automated so they correspond to the mix density, a common trick is to have a shorter snare reverb during the chorus, where the mix calls for more power. As reverbs tend to soften sounds, it might be appropriate to have a longer reverb during the verse. Some engineers have also automated snare reverbs with relation to the note values played – a longer decay for quarter-notes, a shorter decay (or no reverb at all) for sixteenth-notes. It is worth remembering that a longer pre-delay would normally result in a later reverb decay – lengthening the pre-delay and shortening the decay can result in overall more present reverb. Level-wise, many find that a short, loud decay is more effective than a long and quiet one.

Room size

The room size parameter determines the dimensions of the simulated room, and in most cases it is linked to the decay time and the early reflection pattern. Coarse changes to this parameter distinguish between small rooms like bathrooms and large spaces like basketball arenas. Generally speaking, the smaller the room is the more the coloration occurs. Increasing the room size can result in more vigorous early reflections pattern and longer pre-delay – combined with shorter decay time, the resultant reverb can become more pronounced.

Density

A density parameter on a reverb emulator can exist for the early reflections alone, for the reverberation alone or as a unified control for both. The density of the early reflections gives us a hint regarding the size of the room, where denser reflections suggest a smaller room (as sound quickly reflects and re-reflects from nearby surfaces). The density parameter determines how many discrete echoes constitute the early reflections pattern and with low values discrete echoes can be clearly discerned. Reducing the early reflections density can minimize the comb-filtering caused by phase interaction with the direct sound. With both early reflections and reverberation densities, higher settings result in a thicker sound that can smooth the sharp transients of percussive instruments. Low density for percussive instruments can cause an unwanted metallic effect similar to flutter echo. But the same low-density settings can retain clarity when applied to sustained sounds such as pads or vocals (which naturally fill the gaps between the sparse reverb echoes). The density setting also relates to the masking effect – a denser reverb will mask more brutally than a sparse one.

Diffusion

The term diffusion is used to describe the scattering of sound. A properly diffused sound field will benefit from more uniform frequency response and other acoustic qualities that make reverbs more pleasant. Diffusion is determined by many factors, for instance, some materials like bricks diffuse sound more than other materials such as flat metal. An irregularly shaped room also creates more diffused sound field compared to a simple cubical room. When diffusion occurs, the reflection pattern becomes more complex, both in terms of spacing and levels. Many link the diffusion control to density, sometimes in a way that high dif- fusion settings result in growing density over time or produce less regular reflection spacing. Density and diffusion are commonly confused as their effect can be identical.

Frequencies and damping

Frequency treatment can happen at three points along the reverb signal path:

    1. Pre-reverb – where we usually remove unwanted frequencies that can impair the reverb output.
    2. Damping – frequency treatment within the reverb algorithm which relates to the natural properties of the simulated space.
    3. Post-reverb – where we usually EQ the reverb output in order to fit it into the mix.

Pre-reverb equalization usually involves pass or shelving filters. Low-frequency con- tent can produce a long, boomy reverb sound that can clutter and muddy the mix. A HPF placed before the reverb can prevent this by filtering low-frequencies content – like those of a kick in a drum mix. Some high-frequency content might produce a luminous and unpleasant reverb tail – a pre-reverb shelving EQ can resolve such a problem. Damping is concerned with the frequency behavior over time. High frequencies are easily absorbed: It takes 3" (76 mm) of acoustic foam to eliminate all frequencies above 940 Hz; it takes a bass trap 3' deep (1 m) to treat a frequency of 94 Hz. High frequencies are also absorbed by the air itself, especially when traveling long distances in large spaces. The natural reverb of an absorbent space has its high frequencies decaying much faster than low frequencies, resulting in less high-frequency content over time. The damping parameter usually represents the ratio between the reverb decay time and the frequency decay time. More specifically, a standard decay time of 4 seconds and a HF damping ratio of 0.5 means that the high frequencies will decay within 2 seconds. Damping ratio can also be higher than 1, in which case specific frequencies will decay more slowly over time Many digital emulators produce brighter reverbs than those produced by real spaces. Damping HF can attain more natural results or help in simulating a room with many absorptive materials. But HF damping can be useful in other scenarios – the sibilance of vocals can linger on a reverb tail making it more obvious and even disturbing (although often mixing engineers intentionally leave a controlled amount of these lingering high frequencies). Noises caused by synthesized sounds that include FM or ring modulation, recordings that capture air movements like those of a trumpet blow, distortion of any kind, harsh cymbals or even the attack of an acoustic guitar can all linger on a long reverb tail and add unwanted dirt to the mix. In such cases which many novice mixing engineers tend to overlook, HF damping can serve as a remedy. LF damping can be applied in order to simulate rooms with materials that absorb more low frequencies than high frequencies, like wood. Sometimes low-frequency reverberation is required but only in order to give an initial impact. Employing LF damping in such cases will thin the reverb over time and will prevent continuous muddying of the mix. Post-reverb equalization helps tuning the reverb into the mix. After all, a reverb is a long sound that occupies frequency space that might interfere with other signals. Just as relevant is a discussion on how different frequencies can modify our reverb perception, high frequencies give the reverb a spell of presence and sparkle that many choose to retain, especially when reverbs are used as an unnatural effect. On the contrary, a more transparent, even hidden reverb can be obtained if its high frequencies are softened. Very often high frequencies are attenuated in order to create a warm or dark effect – such a reverb can be heard on many mellow guitar solos. Attenuating high frequencies can also result in apparent increase of distance. Low frequencies make reverbs bigger, more impressive and warmer. A boost with a low-shelving EQ can accent these effects. The more low frequencies a reverb has the bigger the space will appear. LF attenuation (or filtering) will thin the reverb and make it less imposing. Reverbs and stereo

Mono reverbs

There are a few aspects which contribute to the limited realism of mono reverbs. One of them is the masking effect, where the direct sound masks the early reflections. An early trick to solve this involved panning the reverb away from the direct sound. But while this can yield an interesting effect, it has very little contribution to realism. The main issue with a mono reverb is that it does not reassemble the directional behavior of a natural reverb, which arrives to our ears from all directions. A reflection pattern that hits the listener from different angles will sound more realistic than one that arrives from a single position in space. Although our pair of monitors can usually only simulate one sixth of the space around us (and even this happens on one dimension only), a substantial improvement is achieved by using a stereo reverb. Mono reverbs are used when some properties of a stereo reverb are unwanted, notably its width. To give one example, say we have two distorted guitars each panned hard to a different extreme, and we want to add a sense of depth to these guitars or just add a bit of life. Sending both guitars to a stereo reverb will fill the stereo panorama with harmonically rich material that might decrease the definition of other instruments. In addition, the localization of the two guitars could blur. Instead of sending both guitars to a stereo reverb, we can send each to a mono reverb and pan both the dry and wet signals to the same position. The mono reverbs might not produce a natural space simulation, but they would still provide some sense of depth and life without filling the stereo panorama. It is worth remembering that usually using a single channel from a stereo reverb will translate better than panning both channels to the same position; this is due to the phase differences that many reverbs have between the two channels.

Stereo reverbs and true stereo reverbs

The implementation of a stereo reverb requires different reflection content on the left and right output channels. It is very common that the two channels are highly phase- shifted, which is why soloing a reverb will often make a phase meter jump to its negative side. These phase differences result in a spacious reverb that spans beyond the physical speakers and can seem to be coming from around us. The drawback of these phase differences is their flimsy mono-compatibility. Some reverbs give different controls that dictate the overall stereo strength – the more stereophonic a reverb is the more spacious it will appear.

Panning stereo reverbs

The practice of panning a stereo reverb return to the extremes is not always justified, since the reflections will occupy the full width of the stereo panorama. The potential masking that a reverb might cause can be reduced if a narrower panning tactic is used. There are many cases where narrowing a stereo reverb is desirable. For example, a strong stereo reverb on a snare can sound foreign to the mix if panned to the extremes. More cohesive results can be achieved if the reverb is panned around the source instrument so it only occupies a slice of the stereo panorama. Identical panning schemes can be applied to nearly every reverb which is used as an effect, including vocal reverbs. In less natural mixes even the ambiance can be narrowed down. This creates a less spacious impression, but more intense effect. It also clears the stereo extremes for other sounds. As narrowing a reverb makes it more monophonic, it can result in a decrease of perceived depth, which can work well in a powerful mix. However, if the ambiance is narrowed too much, it can give the impression that the instruments are located in a long tunnel. Although the narrowing of the reverb output is commonly done using pan pots, these can cause unwanted phase interaction between the left and right channels. It is worth checking if the emulator has a stereo spread control (sometimes called stereo width). With some emulators, using the stereo spread control instead of panning the reverb return can minimize the phase interaction happening between the two channels and produce a healthier reverb altogether. If reverbs are used as a special effect or to fill stereo gaps, they do not have to be panned mirrored around the center. For instance, if a snare is panned to 11:00, it can be reasonable to pan the snare reverb between 10:00 and 12:00

Gated and nonlinear reverbs

The reflections in a real room can build up and decay in amplitude very differently from those produced by a reverb emulator. While most emulators loudest reflection is the first one, the reflection pattern in some spaces can build up for a while before reaching maximum amplitude – an equivalent to the attack stage on a dynamic envelope. In addition, the reflections in a real room fluctuate in level, unlike the linearly decreasing reflection levels in some reverb emulators. The shape of the reverb envelope is important for natural simulation, and some emulators provide related controls. Mixing engineers can employ various tools to make the decay of reverbs far less natural. Despite potentially damaging its natural impression, such a nonlinear reverb can have more impact. The decay of a reverb is most commonly what we are shaping. The simplest way of shaping it is by inserting a compressor after the reverb. Often the compressor is set to moderate threshold, moderate ratio and fast attack. For truly wild decay shapes the release can be set to minimum so some pumping takes place; for more subtle results the release time can be lengthened. The most famous nonlinear reverb is the gated reverb. Rumor has it that it was discovered by accident during a recording session of Peter Gabriel’s third album. It is mostly recognized for its use in Phil Collins’ In the Air Tonight. In its most simple form, gated reverb is achieved by inserting a gate after the reverb and setting the gate threshold so it opens with the initial reverb burst. Usually a fast attack and release are used and hold settings that make rhythmical sense. This configuration does not alter the shape of the decay curve until its later stages. One rationale for gating a reverb in such way has to do with the fact that with sustained dry signals the reverb is likely to be masked by the direct sound as soon as it drops by 15 dB. In a busy mix the reverb tail can be masked by many other sounds – a gated reverb can sharpen the sonic image by cutting these inaudible parts of the reverb. If instead of decaying linearly, the decay shape is altered so it sustains at maximum level (or falls slowly) and then decays abruptly, the reverb becomes more prominent. To achieve this, a compressor is inserted after the reverb. The compressor is set to flatten the decay so it hovers longer at higher amplitudes. But this will result in a reverb that might not drop below the gate’s threshold. Therefore, the original sound is sent to the gate’s key input. There are endless amounts of productions in which gated reverb has been applied on snares and toms, but it can be applied to many other instruments even those of non- percussive nature. Shaping the reverb’s decay envelope can give the reverb more punch or more defined rhythmical characteristics (which must be observed while tweaking the gate). Gated reverbs are so popular that many digital reverb emulators include such a program. Still, better control is usually achieved by setting up the independent building blocks.

Reverse reverbs and pre-verb

A reversed reverb does not really reverse any audio, but it does reverse the amplitude envelope of the reverb so it starts quietly, grows gradually and then cuts abruptly. This effect gives a certain backward impression that can be used as a special effect. A more common practice in mixing is called Preverb. It is done by recording the reverb and then playing it backward so that it rises before the actual sound. In the digital domain, the original material has to be reversed, the reverb has to be bounced and then both the original sound and the reverb have to be reversed again.

Reverbs in practice

Reverb quantity and allocation

A common mixing discussion relates to the amount of reverbs one should use in a mix. First is the fact that it is easier to grow a tree than a forest. Put another way, a mixing engineer might be tempted to use many reverb plugins while compromising on quality and giving less attention to each reverb. Selecting and configuring a reverb can be a truly time-consuming task, especially for the less experienced engineers who are not sonically fluent with their tools. Second, the use of too many reverbs, especially if a different program is loaded on to each, can cause what is known as ambiance collision – a blend of too many spaces, which does not make any spatial sense. A certain objective when using reverbs is the creation of a firm, convincing ambiance even when an imaginary ambiance is involved. Although many reverbs can work together, a certain risk exists if not enough attention is given to how they interact. Generally speaking, when using more than one ambiance reverb, softening the highs of the short reverbs can help concealing them and reduce collision.

Aux send or insert?

Past has it that reverbs, by convention, are fed via auxiliary sends. The most obvious advantage in using a send is that we can route more than one signal to the reverb emulator and therefore save processing power or share high-quality emulators. If many reverbs are used, but each is fed from one single track, the multitude of auxiliary tracks can clutter the mixer, and many buses are employed for sheer routing purposes (rather than summing). Therefore, some argue that in specific situations it is sensible to use an insert rather than a send, for instance, when a reverb is added to a kick during two bars only; or when the reverb only adds a touch of early reflections that are meant to enrich a mono signal.

Pre- or post-fader?

As just discussed, reverbs connected as an insert are pre-fader (with rare exceptions). When connected as an aux send, we have a choice between either a pre- or post-fader send. A post-fader send is also post-cut. Similarly, a pre-fader send is also pre-cut. The basic principle behind using post-fader send is that the level of the dry signal and the reverb level are linked – when the dry signal level increases, the reverb level increases as well. A post-fader send is necessary if any level or mute automation should take place – when we fade out or mute the dry signal we want the reverb to fade out or mute as well.

How loud, how much?

One way to check the effectiveness of reverbs, even if hidden, is by muting them and seeing how they contribute to the mix. Yet, in specific situations having a bold, well-defined reverb is our aim

Tuning ambiance

The standard way to create ambiance involves sending various instruments to one or more ambiance reverbs. It happens that the resultant ambiance suffers from various issues due to the frequency content of the various instruments sent to the reverb. It involves sending to ambiance reverbs a modified version of the instruments sent to the mix bus. On an audio sequencer this can be achieved by duplicating a specific track and disabling its standard output routing to the main mix. Yet, this track is sent to the ambiance reverb instead of the original. We can then process (mainly equalize) the ghost track in different ways, so as to tune it to the ambiance created by the reverb it is sent to.

Delay and reverb coupling

A very common mixing technique involves a blend between a delay and a reverb. Most often this technique is used on vocals, but it can be applied on other instruments just as well. Blending a delay with a reverb is known to result in more impressive effect than having only one of them. One way to look at it is as if the delay enhances the reverb. Most often the delay and the reverb are connected in parallel, i.e., none feeds the other. It also pays trying to connect a delay before a reverb – such an arrangement can also produce an appealing effect.