Welcome to the infomation page about Mixing Music part 2.

The fine art of mixing single audio tracks together as a whole is difficult, specially when you do not have some guidelines. First rule for explaining tthe name 'mixing' is that it stands for mixing it al up together, to make a whole overall sound. This means adjusting overal sound levels and making use of Fader Levels, Panning, EQ, Compression, Reverb, Delay or any kind of effect towards a good balanced track. Several issues come up while mixing, technique and equipment. Also offcourse like in composing, improvisation and goofing around might help you more to understand the difficult task to mix. Important is that the overal mix should be sounding tight and together as one. This Mixing page will try to explain some things about mixing, where to start and how to finish the mixing stage with good results. Remeber that time and understading is the way to go, knowing how to mix is a good thing before starting one. Take a good look around and read the information you find on our mixing information page. When you did not read Basic Mixing I, follow this link.


Basic Mixing II

Mixing a Starter Mix and Static Mix.

In Basic Mixing I, we have explained the starter mix and progression towards a static mix. Basically we have covered dimension 1 and 2 more than dimension 3. We actually did not apply any effects or added anything (that was not there before). The starter mix is aiming for some togetherness and cleaning up what is not needed. Keeping what is needed, with the help of the Level Fader, Balance, EQ and Compression, Gate, Limiter. Without adding effects that work overall on the mix, we try to have all instruments sound best and clear (starter mix), so all can be heard in their range. Together sounding as one mix. To get some headroom back you may have to switch back to the starter mix again. Change one thing can lead to affect the rest of the mix. Keeping track of the mix and dimensions is part of checking and re-rechecking. And should be at constant attention. When a mix starts to be muddy, when two instruments overlap in each other’s frequency range (masking), you will need to correct this by using separation. Remember all instruments are placed inside the frequency spectrum and it is better to spread them out and create some headroom for them to be heard. Actually there are quite some tools for separation. Just as in human talking, as long as only one person is talking to you, you will understand and hear well. When a crowd is talking, it is difficult to understand what is going on. It is a mixing fact that crowding up the mix with more and more sounds is not a good thing. So we actually cut out what is not needed. Making all instruments sound good in their own range is way better (dimensions). Constantly think about how the spectrum will change according to what you add or remove, giving placement in the frequency range for each instrument to shine, but not intrude. Cutting out what is not needed, may clear the way for other instruments to come more upfront. Instead of just boosting, try cutting (other instruments) and make some space. Basic Mixing part I will explain the starter mix, so read this before you go on.

Introducing Dimension 3 (depth).

Basically with applying dimension 3, we will progress with the static mix. (starter mix dimension 3). Here we will go further into adding effects and making the overall sound of the mix, static reference mix. Adding reverb or delay (or any other effect) will add more frequencies and level so is costing some headroom. Effects can affect placement, maybe a stereo delay will get your instrument to move out of its natural position. There are quite a few effects that affect the dimensions and are the tools for basic mixing. Still we separate Fader, Level, Balance, EQ, Compression, Gate and Limiter from the rest, because these are tools that are commonly used. Read basic mixing I for more info on those tools. Also we call the finishing of the dimensions, the finishing of the static mix. Basically the static mix is called by the fact that knobs, faders and settings apply to the whole mix. So we do not automate or place events inside the timeline of the mix, but we just set knobs and faders for the whole mix.



Now, the most interesting, versatile and creative part of mixing, adding effects. Endless effects are provided to create sounds or adjust it. Available are hardware or software effects. We cannot even discuss them all over here, so we only focus on the most used and common ones. Also at first we will focus at effects that work in dimension 3 (depth). Effects are often a welcome addition to a mix, a bit of reverb can do a good job and distortion on a guitar can make it rock. Remember that each time you add an effect; it will change the range and whole frequency spectrum of the mix, possibly gaining frequencies. Therefore filling up headroom more and more. A reverb may add a nice roomy sound, it can also muddy up the mix as a whole. Cutting some lower frequencies out of the reverb signal can help clear up again, especially the 0 Hz to 120 Hz (180 Hz) range. So knowing effects and what they do to the signal is important, keeping in mind what the effect is doing with the three dimensions, quality and reduction, headroom, etc. Just adding effects in a row may sound good at first, maybe later on when your ears are not fatigue, you might think different. Do not rush into adding effects; think what is needed for the mix to get better. Also for most effects we like to cut the lower frequencies, just because they might influence the bass range from 0 Hz to 120 Hz. Be gentle with effects, muddiness and fatigue ears are just around the corner. Because there is a vast amount of effects available, there is no general solution for mixing. We all try to do our best, but we enter the creative field and really are on our own. You can pick up tricks and learn from others, there is a good deal and straightforward information on the net. It can be debated, it can be funny, and it can be good or bad. Everything will stand by how much experience you have with mixing and how much you understand it. Time and learning are again factors of success. Whenever you are tired of not getting what you need out of your mix, be gentle and maybe do a re-check or just stay away for a while and return back later on. Do you really need all those effect to have a good sound? Remember, Less Is More! The more basic approach will work often better and is faster and cleaner. Crowding up the spectrum with effects is never a good idea. The more natural effects are to our ears, the better we can use them to affect the dimensions of the mix.


Track Effects.

Whenever you need a single track or instrument to sound different, you can add a track effect to it. This is common for all kinds of effects. But for single instrumental track effects, fader, level, EQ, Compression, Gating and limiter, are most commonly used for mixing purposes. Keep everything adjustable per instrument or track; this can help even when adjusting the final mix. Track Effects are most common on computers and digital systems, you can place many. But processing power will drop also, it can be rewarding to separate things and keep effects to a minimum. Less is more.

Send Effects.

On analogue mixers Send Effects might be all that you have; digital systems do have send effects also. Whenever you need an effect that works overall on several instruments, you can use the send effect and send the signal to the Send Effect. Most likely the Return of the effect will come up on the Master Fader. For send effects you can be efficient with processing power, because using only one instance of the effect for multiple instruments or tracks. Also it can be fun routing send effects and be creative with sound and effects, trying effects after each other before deciding what works best. Send effects are effective as a collective on all instruments. Having two or more send effect channels can help layer the mix, but we try to stay away from send effects when we could use groups instead.

Group Effects.

As we did layer instruments and grouped them, we gave each layer of our mix a separate group track. A compressor for welding purposes or an EQ could be placed on a group track. As opposite to a group track, send effects can be uneasy to keep track of. Thus meaning the routing of the send effects can be inputted from different tracks or instruments, sometimes this can work confusing. Place effects on group tracks, when you can. Else use a Send Effect track. Especially when you apply each time the same effect on separated group tracks (repeated instances of the same reverb), you could choose to just use one instance on a send track.

Pre-Fader and Post-Fader.

The option to place effects pre-fader or post-fader is a matter of purpose. Any effect placed pre-fader as an insert is affecting the track signal before the fader level, balance, etc is applied. For track compression on vocal for instance, we mainly use a pre-fader compressor. This way the threshold setting of the compressor is not affected by the fader setting of the track. We can now adjust the level of the vocals with the same kind amount of compression. If we place the compressor post-fader, the threshold is affected by the setting of our track fader (even balance, etc), so the amount of reduction is influenced. For vocal we choose pre-fader compression, this way the amount of reduction stays the same when we adjust the single track. But the same kind of system is applied to all other effects we place inside the mix. Placing pre-fader means, the signal will first be affected by the effects in place, second by the track mixer settings (fader, balance, pan, gain, etc). Placing post-fader means, first the track settings and then the effects. Post fader effects are for instance reverb, delay, echo and all other sound manipulation effects alike chorus, phasing, modulation, etc.

Our Stage Plan according to dimension 3 (depth).

We have discussed and applied the first dimension (panorama) and second dimension (frequency spectrum) for getting a good starter mix. Now as we would like to finish off all dimensions according to our stage plan, we can apply some depth in dimension 3. Mixing all dimensions is called a static mix. Mostly we are talking reverberation sounds that influence our hearing in perceiving depth, as we have finished dimension 1 and 2 (2D), dimension 3 should be our first concern. In dimension 1, we have set panorama. In dimension 2 we have set frequency range. By rolling off some trebles or highs, we could affect distance for dimension 3. But however dimension 3 is mainly a reverberation effect that will make our ears believe there is some room or distance. Suddenly the field becomes 3D with all dimensions in place.



Our hearing can calculate or guess the distance (depth) by hearing the dry signal and its reverberations. Especially the pre-delay between dry signal and first reverberation makes us perceive depth. Reverberations occur when a dry sound is hitting solid objects alike walls or any other objects placed into a room. Even outside objects like water, mountains, valleys, tunnels, ambience, etc, somehow cause reverberation to be transmitted back to us (echo). Specially calculation of the time between the dry signal (0 ms) and the first reverberation signals ( > 0 ms) come across (returning to our ears a bit later) is making our brains understand depth or perceive distance. Pre-delay is an important factor for any delay or reverb effect to be taken in account while we are aiming for depth or distance inside dimension 3. Because the first transients of any sound will make our brains react to recognize and understand, this goes for the dry sound as well as for the reverberation sounds (any sound). The most used effects for perceiving depth or distance are reverb and delay. We also explained before that dimension 2 (frequency range) by rolling off trebles or higher frequencies we can perceive the dry-signal to be distanced. Depth means distance. When we are using dimension 1 (panorama), when we placed a dry signal more to the left, the left speaker will play more than the right speaker does. But with reverberation in dimension 3, for perceiving depth as a room, we must transmit the 3D Spatial Information to the listener. We could place this at the opposite side on the right. When a dry signal is playing a note from start, the reverberations returning from the room slightly later in time (specially the first pre-delay), make our brains understand and calculate some kind of distance (depth). In combination with panorama (dimension 1), we can use dimension 3 (and 2) for applying our stage plan. Apart from using treble roll off or high frequency roll off in dimension 2 to perceive depth, common used effects for dimension 3 are Reverb and Delay. Apart from creative aspects we will discuss later on, we will use reverb or delay to represent the dry instrument (transients, Sustain) in our stage plan, tracks or mix with some more natural perceived acoustics. This is called 3D Spatial Information, the information needed to make our hearing and brains believe in depth or distance.


You get very different results from your filtering depending on where you put the filter in the signal chain. To introduce some real movement into delay lines, for example, place sweeping low-pass filters before the delay. You then get the movement of the dry sound contrasting with the movement in the delay line. If both the filter sweep and the delay lines are tempo-sync'd, you can create interesting effects where the filter appears to be moving up and down at the same time. Filters are also great for use on drum loops. One trick I like is to send the drums to a modulated resonant filter set up as a send effect, with a narrow band-pass EQ beforehand. This creates a rather bizarre metallic melody that accompanies your drums. It can get fatiguing if over-used, but brought in at a low level in some sections of a song, it can create plenty of interest, particularly if followed by a modulated delay. Delay is a most simple effect and will repeat the dry input signal after a certain delay time. Basically delay is a kind of reverberation, although less overcrowding then a reverb, using lesser reflections. A delay does not often represent a room, but simply delays the dry signal until the first delay is reflected (repeated). The delayed signal may either be played back multiple times, or fed back in to the input signal (feedback), to create the sound of a repeating, decaying echo. The first delay effects were achieved using tape loops. With some feedback the delay effect can be more exiting. In Reggae the echo effect or delay is used in various ways but feedback is important for creating that 'dub' effect. Delay's and gates are often synced to tempo of the track.

Mostly the delay signal will have some kind of ADSR time settings, so the delay signal will fade out in time. A delay becomes interesting at a faster tempo and will prevent usage of a reverb instead. A delay is less muddy and fuzzy compared to a reverb, so mostly a delay will likely keep instruments upfront. In these modern times the delay can be in sync with the tempo on beats and bars. An early version is the Multi Tap Delay, see below.

Sometimes the delay has a step-sequencer or matrix, nice settings are 3/16 and 8/16. Delays will come in various shapes and sizes, discussing them all would be a hassle. But in general mixing and for improving sound on separate instruments, delay is a common used effect. Most times a delay is used as a creative tool, but can also be used for perceiving depth. As a start it is better to use a delay, instead of using a reverb. Sometimes you can create a clearer reverb effect using a delay and some creative settings. Remember that a delay (specially the delayed return of the dry signal) can be perceived as depth or distance (dimension 3). Delay will leave more headroom then reverb and will sound more open. A Ping-Pong delay is a crossed over delay and combines left and right signals, see below.

A Ping-Pong delay or stereo delay can affect the panorama, this can affect the dimensions. Watch out for these kinds of stereo effects and only use when you need it. A Ping-Pong delay or stereo delay can be creative, but however also can avoid masking. Temporarily unmasking by swaying the automated stereo delay. The trick with mixing delay is setting up inside a mix, most likely to adjust the delay to the point you don't really hear it, but it is there. For main vocals that must stay upfront, we could use a delay, keeping original main vocals ear able and still have the ambient early reflections. Use a gate to control what is passed into the delay or goes out from the delay, this will separate the delay effect even more and will not be a mix filler. To prevent muddiness use EQ to cut the lower bottom end from 0 Hz to 120 Hz (180 Hz). Delay is common used as send effect and less as track effect. So the ultimate place would be on a send or group. Remember for perceiving depth or distance we need the dry signal to be heard unaffected, on top of the dry signal will sound the delay. We can roll of some highs to make some more distance or depth. When you need an instrument to sound upfront, you keep the high trebles in place and use no pre-delay or little. When you need an instrument to sound distanced, roll of some high trebles and use more pre-delay. Conflicting signals inside the 3D Spatial Information mean our brains will be confused. Rolling off highs for distance and then set no pre-delay is sending conflicting information. The natural world sound our hearing likes so much, is sometimes uneasy to recreate while mixing.

Tempo Delay: Most plug-in and hardware delays now allow you to automatically sync delay times to MIDI clock and then specify the interval of the repeats in terms of note values rather than milliseconds. A trick here is to use two simultaneous tempo-based delays with, say, a triplet delay setting, panned hard left, and a straight-note delay panned hard right. Things can get more interesting still if you apply this technique using ping-pong delays, so that alternate repeats bounce from one side of the stereo spectrum to the other. To create a true 3D effect, play around with the amount of original signal left in the middle. Depending on the intervals between your repeats, you can turn simple guitar and synth lines into complex, arpeggiator-like patterns or totally spaced out ambient pieces. Stephen Bennett
Ostentatious Delays: If you're making very rhythmic music of any kind, it makes sense to use tempo-sync'd delays, to avoid undermining the main pulse. However, simple tempo-sync'ed delays tend to be masked by the main rhythmic stresses, so they sink into the background of the mix unless mixed very high in level, which makes it difficult to create ostentatious delay effects in rhythmic music without swamping your mix. One solution to this problem, very common in trance music, is to set a delay to a three-16th-note duration, which means that although the delay repeats never step outside the 16th-note grid, they'll often miss the main beats and therefore remain clearly audible.

Keep It Reel: Perhaps because a humble tape echo was the first effect I ever owned, delay has always been my primary effect. Whether to liven up repetitive loops or add apparent complexity to simple solos, it's worth getting to grips with delay the old-fashioned way. This means daring to switch off MIDI sync and manually setting delay time, driving feedback to the brink of madness, or routing the pure delay output through equalisers, filters and so on. Many of today's digital delays allow you to darken the delay iterations, but there's no reason not to find your own method to achieve this: adding alternative colours and discovering your own favourite processes. I find precise, perfect digital delays can be rather generic and characterless — so the more I delve into additional treatments, the more interesting and organic the results are.

Softer Delays: I'll usually have at least a couple of delays as auxiliary effects in a rock or pop mix, but I often find that bringing the general level of the delay as high as I want it makes any transients stand out too much. When I'm sending single notes on a clean electric guitar to a delay line, say, I tend to want to hear a wash of sound, not the rhythmic 'CHA-Cha-cha-cha-cha' of a repeated note attack. For this reason, I'll often put a gate or expander before a delay, with an attack time set to 10ms or so. This is enough to 'chop off' any abrupt transients, and makes the delay sound much smoother. Sam Inglis
Non-sync'd Delay: We are so used to perfectly sync'd delays that it's easy to forget that manual sync and a pair of ears has a charm all of its own. Even delay times that bear no obvious relationship to the tempo can add dynamic movement and feel to a track: check out some early King Tubby if you need reminding of this.

Subtlety: You don't always have to make longer echo or delay effects obvious in the mix for them to be effective. Once you've set up the delay times and panned them to suit your song, try dropping the delay levels until you scarcely notice them during most of the mix (listening on headphones often helps set the most suitable level). This generally results in intriguing little ripples of repeats that you notice at the end of verses or during pauses, that add interest and low-level detail to the mix. 

Calculating Delay to tempo.

Delay mainly affects tempo. Thus percussive instruments (drums) that need their existence to be heard rhythmically, need to be in sync with the tempo. You can fiddle around until you find a good setting, but also calculation of the delay time will give you a hint where to start. Tempo Delay, calculate 60 ÷ BPM = the delay time of one bar in seconds. Divide this by four to give the delay time per crotchet. Divide this by 1000 to convert seconds to milliseconds. When you know the BPM of a mix we can calculate the delay time with:

60000 / BPM = delay time in ms.

Or for any kind of note:

(60 / tempo in BPM) * 1000 ms * 0,75 (dotted quaver)
(60 / tempo in BPM) * 1000 ms * 2 (half note)
(60 / tempo in BPM) * 1000 ms * 0,666 (crotched triplet)

It is good to have some control when mixing, so a separate controller could help to live mix the delay or any effect (this we will discuss in Dynamic Mixing). Long delay times can be recognized by the brain as echo. Short delay times can be recognized as ambient or psycho acoustic (small room reverb, ambient) and can affect the spread of the sound (depth or distance). Reverse delay or backwards echo, is a reversed sample played backwards with an added delay. Then reversed again. For Reggae Dub Delay, use a single delay return. Feedback the delay output to itself. The aux send can be used real-time (or with automation, dynamic mixing) to dub over the original sound. (Boost some EQ around 3Khz and roll of some highs and lows for Dub).

The most familiar use of delay processors is by guitarists in popular music, employing delay as a means to produce densely overlaid textures in rhythms complementary to the tempo/sync of the overall piece (this is a creative aspect). Electronic musicians (synth, sampling) use delay for similar effects, and less frequently, vocalists and other instrumentalists use it to add a dense or ethereal quality to their playing (without pushing them back rows on the stage, keeping it more upfront compared to reverb. Extremely long delays > 10 seconds or more are often used to create loops of a whole musical phrase. Sometimes unsynced to tempo delay is used for a solo instrument (playing a solo for a while and then return to normal song/static mix reference level).

Echoplex is a term often applied to the use of multiple echoes which recur in approximate synchronization with a musical rhythm, so that the notes played combine and recombine in interesting ways. On computers or digital systems this can be achieved by a step sequencer or matrix.

Doubling echo is produced by adding a short range delay to a recorded sound. Delays of 30 ms to 50 ms milliseconds are the most common. Longer delay times become slap back echo, sync them to tempo. Mixing the original and delayed sounds creates an effect similar to double tracking or unison performance.

Slap back echo uses a longer delay time (75 ms to 250 ms), with little or no feedback. The effect is characteristic of vocals on 1950 Rock and Roll records, particularly those issued by Sun Studio. It is also sometimes used on instruments, particularly Drums and Percussion. Slap back was often produced by re-feeding the output signal from the playback head of a tape recorder to its record head, the physical space between heads, the speed of the tape, and the chosen volume being the main controlling factors. Analog and later digital delay machines are also easily producing the effect. Slap back delay between 20 to 80ms, no feedback. Sync to tempo to make rhythmical correct.

Flanging, Chorus and Reverberation are all delay-based sound effects. With flanging and chorus, the delay time is very short and usually modulated. With reverberation there are multiple delays and feedback so that individual echoes are blurred together, recreating the sound of an acoustic space.

In audio reinforcement a very short delay often of only a few milliseconds, is used to compensate for the relatively slow passage of sound across a large venue. The unmodified signal is not played, and the delayed signal is set to leave the speakers at the same time or slightly later than the sounds passing from the stage. This technique allows audio engineers to use additional speaker systems placed away from the stage, but give the illusion that all sound originated from the stage. The purpose is to deliver sufficient sound volume to the back of the venue without resorting to excessive sound volumes of a large sound system placed near the stage exclusively.

A delay tail on the front vocals, make the vocals appear with more warmth and appear fuller. Without putting the frontal placement into jeopardy. The more the delay is appearing in the mix, the more it will cover the vocals, using ducking on the first part of the vocals can free up fuzziness. Delays should only be used as a creative event but can give certain distance. Certainly even artistic is a Band Echo combined with a Spring Reverb, this is called dub. It is also common to give the main vocals a bit of ambient reverb (small room, drum booth) after the delay, to have some more togetherness with the rest of the mix).

Delays generate space, tempo synched.  Divide 60000 ms (one minute) by the song tempo (quarter notes per minute) = ms.  Variations in delay time drive (shorter) or drag (longer) the rhythmic feeling, we could use automation but that will be after we finish the static mix. Avoid phasing <  10 ms. Single delays 10 and 30 ms long thicken up sound, while the original sound is localized (upfront), perceived single delays  as direct sound events (early reflections).  Stereo delays are  suited for rich sound with a low level room / ambience effect. Delays between 30 and 60 ms are called doubling effects (Beatles). Delays between 60 and 100 ms are slap echo (Elvis). Stereo delays < 100 ms are acoustic space.  > 100 ms is echo distance and space. The longer the delay time, the more sound appears to be indirect. Delay tends to blur sound less than reverb. To discretely create space, delay should be very subtly used, so that you miss the muted FX channel, but don’t really perceive the delay when turned back on. Echo, longer than > 100ms which is not tempo synced is good for creating an effect that is clearly heard as such in the mix (solo’s).



To simulate the effect of reverberation in a large hall or cavern, one or several delayed signals are added to the original signal. To be perceived as echo, the pre-delay has to be > 50 ms. Short of actually playing a sound in the desired environment, the effect of echo can be implemented using either digital or analog methods. Analog echo effects are implemented using Tape Delays (Band Echo) or Spring Reverb. When large numbers of delayed signals are mixed over several seconds, the resulting sound has the effect of being presented in a large room, and it is more commonly called reverberation or reverb for short. Reverse Echo is a swelling effect created by reversing an audio signal and recording echo or delay whilst the signal runs in reverse. When played back forwards again the last echoes are heard before the effected sound creating a rush like swell preceding and during playback. Jimmy Page of Led Zeppelin claims to be the inventor of this effect which can be heard in the bridge of Whole Lotta Love. An echo is a reflection of sound, arriving at the listener some times after the direct sound (early reflections). Typical examples are the echo produced by the bottom of a well, by a building, or by the walls of an enclosed room. A true echo is a single reflection of the sound source (dry signal). The delay time is the extra distance divided by the speed of sound (pre-delay). If so many reflections arrive at a listener that they are unable to distinguish between them, the proper term is reverberation. An echo can be explained as a wave that has been reflected by a discontinuity in the propagation medium, and returns with sufficient magnitude and delay to be perceived. Echoes are reflected back from walls or hard surfaces like mountains. When dealing with audible frequencies, the human ear cannot distinguish an echo from the original sound if the delay is less than 1/20 of a second (50 ms >). Thus, since the velocity of sound is approximately 343 m/s at a normal room temperature of about 20°C, the reflecting object must be more than 16.2 meters away from the sound source for an echo to be heard by a person. Signals that return before < 50 ms are perceived as more ambient. Between 100 ms to 300 ms with some feedback.

Echo and Delay.

Echo and delay are created by copying the original signal in some way, then replaying it a short time later. There's no exact natural counterpart, though the strong reflections sometimes heard in valleys or tunnels appear as reasonably distinct echoes. Early echo units were based on tape loops, before analogue charge-coupled devices eliminated the need for moving parts. Today, most delay units are digital, but they often include controls to help them emulate the characteristics of the early tape units, including distortion and low-pass filtering in the delay path and pitch modulation to emulate the wow and flutter of a well-used tape transport. While pure digital delay produces perfect echoes, an analogue emulation can be more musically useful, as each successive echo becomes less distinct, creating a sense of distance and perspective. Hi-fi echoes tend to confuse the original sound, while the human hearing system seems better able to separate lo-fi echoes from the original clean sound. The feedback control regulates the number of echoes by feeding some of the output back to the input. If you apply too much feedback the delay unit will self-oscillate — an effect often used in dub music. Delay normally relates to a setting with no feedback whereas echo uses feedback to produce a series of diminishing repeats. You don't have to use long, distinct delays: short delays up to 120ms can be used to create vocal doubling effects, normally set with little or no feedback. Nor do you have to dedicate a delay to a single sound: you can configure it via an aux send so that several tracks can be treated with different amounts of the same delay or echo treatment, which not only saves on processing power (or buying separate units!), but can help to make elements of your mix work better together. You can often use a tap-tempo or tempo sync facility to get your echoes exactly in time with the song if that's the effect you need, but many echo/delay plug-ins can be locked to your sequencer's master tempo, enabling you to create precise, rhythmic delay effects.

Modulated Delay

Though modulated delays are essentially effects, the need to balance the dry and delayed sounds as a means of regulating the effect strength means that using these devices via insert points makes them much more controllable than trying to use them in an effects send/return loop. If you do use them as a send effect, you can achieve this balance by automating the send level.


Reverb stands for Reverberation or reflections of sound hitting an object. The normal objects are Walls, Floor and Ceiling. But all objects that reflect the dry signal back to the listener are reverb signals. A Reverb does often represent a room, hall, booth, cavern, cathedral or is ambient. A reverb can transmit more reflections then a delay. Therefore a reverb can be easily overcrowding the mix. Deep sounds have more energy than high sounds. High frequencies loose more levels then lower frequencies over the same distance. There are three areas of reverb perception. Firstly, there's the whole issue of an appealing (a good natural reverb sound). Second, the sense of distance (depth, pre-delay), which is influenced by the dry signal (direct sound energy, transients) and the start of the early reflections from the reverb (or room). Reflections in nearly any time frame will cause a feeling that you are at some distance from the originating sound. This distance effect will be made up of original direct sound and its relationship to duplicate delays. The direction of the echo or early reflections is also important and must be placed naturally accepted to our ears (dimension 3). We could use a nice roll off on the trebles to set some more distance (dimension 2). Reverberation. Cut high frequencies before using reverb. Use true stereo reverb for placement, expansion of panorama. Use pre-fader or post-fader.

think it's fair to say that we all have a pretty good idea of what reverb is, though there are several ways of emulating it in the studio. Early reverb chambers, plates and springs have now given way to digital solutions, which fall into two main camps: synthetic and convolution. Synthetic reverbs take an algorithmic approach, setting up multiple delays, filters and feedback paths to create a dense reverberation effect similar to what you might hear in a large room. Though these often sound a bit 'larger than life', they've been used on so many hit records that we now tend to accept their sound as being the 'correct' one for pop music production. Most can approximate the sound of rooms, halls, plates and chambers, but in comparison with a real reverberant environment, the early reflections often seem to be too pronounced. The advantage of a synthetic reverb is that the designer can give the user plenty of controls for altering the apparent room size, brightness, decay time and so on. In recent years, convolution reverbs have become both affordable and commonplace. These differ from synthetic reverbs insomuch as they work from impulse responses (or IRs), recorded in real spaces to faithfully recreate the ambience at the microphone's position when the IR was made. Sometimes these are referred to as sampling reverbs but there's no sampling involved as such, even though the process seems akin to sampling the sonic signature of a room, hall or other space. Because IRs can be recorded in virtually any space, convolution reverbs generally come with a library of IRs ranging from small live rooms to famous venues, top studio rooms, forests, canyons, railway stations and just about anything else you can think of. They sound very convincing, and there's plenty of variety to be had, but once the IR is loaded, there's only a limited amount of editing you can do without spoiling the natural sound. Usually you can apply EQ and also change the envelope of the reverb decay to make it shorter, and adding pre-delay is not a problem, but after that you pretty much have to take what you get. Some companies, such as Waves, have managed to create additional controls but, as a rule, the further you move from the original IR, the less natural the end result. Ironically, the sound of certain synthetic reverbs is now such an established part of music history that most convolution reverbs come with some IRs taken from existing hardware reverb units or from old mechanical reverb plates. Also, if you have a convolution reverb, it is worth checking the manufacturer's site, as additional IRs are frequently available for download. All serious reverb units have a stereo output to emulate the way sound behaves in a real space and, in the case of convolution models, the IRs are often recorded in stereo, using two microphones. Some surround reverbs are also available. Reverb creates a sense of space, but it also increases the perception of distance. If you need something to appear at the front of a mix, a short, bright reverb may be more appropriate than a long, warm reverb, which will have the effect of pushing the sound into the background. If you need to make the reverb sound 'bigger', a pre-delay (a gap between the dry and wet signals) of up to 120ms can help to do this without pushing the sound too far back, or obscuring it. Though reverb increases the sense of stereo width, it dilutes the sense of stereo position. If you want to pinpoint the placement of something in a mix, you should consider using a mono rather than a stereo reverb, and panning this to the same place as the dry sound. Most synthetic reverbs allow you to balance the level of the early reflections and the later, more dense reverb tail. If you want to keep the sense of space but without the reverb tail taking up too much space in your mix, you can increase the early reflection level and reduce the tail level. As a rule, you don't add much, if any, reverb to low-frequency sounds, such as bass guitar or kick drums. Where you need to add reverb to these sources, short ambient space emulations usually work better than big washy reverbs, which tend to make things sound muddy. Taking this a step further, you can also make a mix sound less congested by EQ'ing some low end out of your reverbs.

Quality of reverberation.

Go through your available reverbs, examine them all. A reverb may sound good while playing solo; it might be bad sounding when you hear the whole mix. If bad reverbs have weak stage depth in the final mix result, they will sound fuzzy or muddy. Bad reverbs need a lot of reverberation power inside the mix to transmit the 3D Spatial Information to our listening ears.

Good reverb can be perceived as depth to a listener (stage depth). A bad reverb is less effective in perceiving depth and have to be set louder, thus can muddy or fuzz our mix faster the a good reverb. Test your reverbs with a drum booth preset (ambient) and a dry drum track. Sort your reverbs out which ones are best. If you have a reverb that sound naturally good and when switched of is making the drums flat again. Then you have a good reverb! Write down for later use, because when you need 3D Spatial Information inside your mix, you would not like to go through all reverbs to find a good one (or just have a bad decision made by planning a bad reverb). It is a timesaver when you already know what reverbs will sound best and can be used in other mixes as well. It is likely that on digital systems now days Impulse Response Reverbs are a good way of transferring 3D Spatial Information. With a good deal of naturally sampled rooms and ambiences the impulse response reverb is most naturally sounding and gives depth in most cases without adding too much mud or fuzziness. Combined in the mix with an algorithmic reverb (based on calculations only) can be a good solution for balancing processing power and quality of sound distribution. But however you must never stay in the mix with a dull sounding reverb that does not add anything and does not transfer the 3D Spatial Information that you need. It is crucial to know what reverb is about, so this can take quite some time to accomplish. Overdoing the reverb is a common beginner’s problem; try setting the reverb level as you think it should be, and then reducing the reverb level by 4 to 5 dB. The masking effect applies to effects as well as the original signal, unmasking a reverb path can make you need less reverb level and have a cleared pathway and leave more headroom / dynamics. If confused, write down the delay or reverb (depth) pathways into your stage plan or pre-plan this whole subject. Masking is always there, but as much as we can reduce it from happening, is a better goal then just boosting and raising levels.

On a digital system, there is need for processing power for a good reverb to shine. The calculations needed for a good reverb are immense. So do checkout all your reverbs in the mix, a good reverb pays off in the stage depth and can be heard at lower levels. They can transmit the 3D Spatial Information without the need of overpowering the mix and will create more depth or distance, with less power be persuasive. Just add a touch of good reverb and you will notice you will need less power to transmit the 3D Spatial Information with the right kind of reverb. Even a good reverb must be set higher in level then you might naturally want to, just to transfer the acoustics to your ears. The acoustics or 3D Spatial Information contains the information of the dry signal and its reverberations and therefore let us perceive distance and depth (dimension 3). This is only accomplished by forcing the 3d spatial information on to the listener’s ears. When you have a low level reverb setting, then maybe you can't hear the 3d spatial information correctly and falls behind in the mix (masking). Then apply more to perceive and push to reverb to a higher level. With a good quality correctly chosen reverb you are able to have enough reverb to transfer the 3d spatial information and still are not flooding the mix with reverb (fuzz or mud, masking, unmasking). Best is to switch from dry mix to reverb and repeat this a few times (while listening the whole mix), adjust the reverb level until your happy with the combination of dry and with the reverb on top of it. When the reverb is sounding muddy or fuzzy by doing this, either choose another reverb or EQ the reverb or remove some low frequencies (0 Hz -120 Hz, > 180 Hz or even more). Muddiness can be easily avoided by EQ, but maybe you can find good quality presets for different kind of purposes, wisely chosen reverb that will just work better and produce less muddiness or fuzziness.

Reverb is a sound that returns all unlimited reflections of a room (or any object in its path), from all directions and distances at various levels. These reflections can be extremely lower in level (-70 dB to -90 dB) compared to the dry input signal. But nevertheless the listener will perceive the 3d spatial information and can guess some kind of distance or depth. Even if noise is added, the special information is still there. But however we try to keep noise away from the 3d spatial information. Basically the dry signal (specially the transients) must come through unaffected, so the listener can hear the transients and measure distance by hearing the upcoming reverberation (pre-delay). If a delay arrives within < 15 ms of the original source signal it will create imaging or panorama problems for example, if you have a sound panned in center and a delay of 1 ms to 15 ms on the right, what you will hear is the image in the center, shifting to the left. This is caused by the characteristics of human hearing in its relationship to localization. The ear perceives localization because a sound wave will arrive at one ear slightly later than the other ear, as part of length of travel. This is an innate survival mechanism for human behavior. It is otherwise known as the Haas effect. If a delay of 1 ms to 15ms is brought back and panned to the same position as the original you will create phasing effects. Also our hearing can perceive louder signals or softer signals being distanced. If a delay signal arrives later than 15 ms but before 100 ms (approx.) it will create more depth or distance (dimension 3). For what you have done is alerted your psycho-aural response, which tells you that you are listening to the sound in a reflective environment, now our brain can guess the distance better. Whereas if you just heard the original dry sound (transients), only the psycho-aural response (reverberation transients) would create the effect that you are standing in a field (panorama and depth together). Our stage plan is based on dimension 1 and 3 most the most part. But however rolling off some highs for distanced instruments or tracks in dimension 2, can help perceive distance better. Dimension 2 can be used on the dry signal and also on its effects (reverberation's). Where most mixes really are going wrong is a unwisely (according to stage planning, maybe conflicting information) chosen reverb. Also contradictory information, alike using a large reverb with lots of highs.

Most people can imagine a sound in a church ,cathedral, a large hall, etc. Most of that sound is just natural events, originated from nature, what we hear from our own world in real life. When we mix a piece of music together it will soon sound dry, unnatural. Reverb is a tool to add nature and make the listener feel at home. Flat mixes tend to have none of the natural reverb, so we can try to add some reverb (or other effects) to make a more natural sound (ambience). Mostly when people have a reverb in hardware or as a plugin, they will tend to search for a reverb that sounds 'good' . With this method it can be time consuming, but worthwhile to examine your effects, try them out extensively and come up with a good list of presets. You can try to make a dry trumpet sound natural by using a reverb and searching for a suitable sound, better is to keep it all correct to natural hearing laws. Some people can imagine in their head how things will sound in their natural context. Stage planning in the 3 dimensions is important, but when it comes down to creating sounds in that 3d environment, imagination can be a helpful tool. Maybe the trumpet sounds best in the place where you can imagine it to be. Then maybe you can select a suitable preset (alike a large hall preset) faster, maybe fiddle a bit with controls to make the large hall suitable. Anyway some people just hunt down for suitable presets, some think before starting off and imagine how it might sound and make a stage plan, then take the preset most suitable straight away. Reverb or delay can enhance the natural sound of your mix. Every element of the 3 dimensions alike volume, panning, EQ, compression, depth, etc, can be seen as an element to control toward a natural sound. Alike Delay, Reverb is a tool to control depth. Other effects alike flanging and phasing, are more unnatural sounds. Effects are nice, but remember to know what purpose you are using effects for. Mostly I think a natural sound is better than a completely dry sound (so all sound could at least use a small reverb or ambience, depth in the form of natural reverb's or delays / early reflections are what we hear 99% in our daily lives. How minimal in a mix used, natural depth will ease the users mind and is likely to be better. However how easy reverb and delay can be setup, it will always be difficult to mimic the natural world.

Basic Reverb rules.

When using more than one reverb, organize by room size. Reverb tends to blur the mix more than delay. The balance between space and distance can be controlled with the effect level. Reverb length particularly with gated reverbs and snare reverbs should be tempo synced. Snare reverb tends to end on the next full beat. Reverb with very short decay times creates discrete places. The longer the delay time, the earlier distance is created (along with the level). Rich trebles content indicates nearness, lack is distance. The main ambience usually drums ambience should be discretely mixed. The problem selecting presets is that reverb never is to be listened in solo mode, but always must be listened and selected in full mix mode. For instruments that are not fundamental (maybe some fundamental ones) place the original dry signal left and the reverb signal right, or vice versa. Test a good reverb on a whole mix and see if it still stands out, you cannot judge a good reverb solo. Take a dry drum group and setup an Ambience or Small Booth. Switch the reverb on and off, a good reverb does not need to be very loud, but you should miss it when turned off. If the reverb sounds natural. then you have an excellent reverb preset or device. 

Reverb Controls.

Pre-delay - The distance in time between the onset of the original sound and the beginning of the reverberation sound expressed in milliseconds (ms). Pre-delay is an important parameter to set the distance or depth (dimension 3). Pre-delay here, is the time span from direct sound to the first reflections added by the reverb. The longer the time spans the greater the distance of the sound source from the listener. Pre-delay with percussive instruments (drums) must be used with great caution. For all percussion instruments including drums and bass, use no pre-delay or up to < 10ms, checking rhythmic (we can use an high trebles roll off for setting distance). High pre-delay for choirs and strings can send them assigned to the back, to the back rows (stage planning). Pre-delay is setting the distance in reverb. High delay times suggest closer, but is more fluttery and less tight. Pre-delay between 50 and 100 ms is sloppy when not synced/, drums and bass should have reverb without delay or very low 0-10ms, if needed longer (don't) and synched to tempo (always). All sounds relating to rhythm, such as drums and bass, should have reverb without almost to none pre-delay. Up to 10ms. Check rhythmic consistency. High pre-delay up to 60ms are good for chorus and strings, to put them in the back of the stage. Pre-delay with very acoustic natural mixes, follow the natural behavior of pre-delay; longer delay times for nearby and shorter for far away. In pop music, use the opposite approach; short delay for nearby and long for far away. With percussive instruments use short delay times or sync to the rhythm/tempo. If reverb muddies up the dry signal, try a higher pre-delay value, sync it. The quality of the early reflections of a reverb is important, only use the best plugins of reverbs and delay/reflections. A reverb may sound good in large hall or big rooms / stadiums, when early reflections or ambience is at hand, the reverb may fail. Good reverb is half the work and means a good start, keep track of reverb presets for later use. Instruments that tend to be close or upfront only use small reverbs or room/booth ambience reverbs, keeping the trebles alive (don't cut), have no fuzziness or blur, so that the early reflections (or the transients of the reverb) are clear to hear. Instruments that are more backward can use larger spaced reverbs and duller ones.

Deep sounds have more energy than high sounds. High frequencies lose more level over the same distance. The greater the distance between the listener and the sound event, the lower proportion of high frequencies in the reverb signal. This is why treble-roll off of the reverb signal is one of the most effective psycho acoustic means of representing distance to a sound source, since our ears intrepid this information subconsciously. Reverb should have more treble for close sounds in front of the mix, away less trebles for the back.

Decay Time - The length of time from the onset of sound after the initial sound has been established until it has dropped in level by 60 db.

Diffusion - If the diffusion is set to high (reflections very close in time) it will make the reverb sound very smooth. If it is low you might start to hear discrete delays that might clutter the sound.

Room size - The larger the number, the bigger the size of the reverb space, the bigger the room is perceived. Some preset programs will introduce more early reflections into the reverb algorithm.

Modulation Rate and Depth - Randomly shifts the time and intensity of the early reflections, creating a more authentic effect. If using a lot of this function you need to be aware of any pitch variances of signals with a lot of harmonic content.

Density -The amount of first reflections, early reflections and the time difference between them. You also have control over the amount of this effect in the reverb mix. Often used for creating good room sounds for drums.

Frequency Controls - All reverb loses high frequency content over time. If you EQ a lot of high-end over the diffused part of the reverb it tends to sound very unrealistic (use quality EQ or oversampling EQ). In most Plate and Hall algorithms the high frequency response gradually tapers off over time. There are also frequency level controls at various low frequencies to keep the reverb from sounding muddy.

Reverb is generally used as a group effect or send effect and sometimes used as an insert effect (this way keeping the dry signal intact). It is likely to place the reverb post fader, or else the fader movements will affect the amount of reverb. By this pathway, when you move the fader the reverb will be the same amount all-time. Set the reverb mix or ratio to 100%, because we already have the dry signal heard (reverb placed as send effect), we do not have to mix the dry signal inside the reverb again. Pre-delay and frequency range of the reverb signal can be perceived as depth. Test your mix at low levels and see if the reverb still is effective, listening reverb or 3d spatial information at high levels can perceive better (but also can fatigue your ears). But your mix must be in place when listened at softer levels also. A well 3D (three dimensional) unmasked mix stands when played at all levels. A good reverb does not need to be so much audible as a bad one, but you will miss it when it is muted from the mix. The treble roll off of a reverb signal is the most powerful way to perceive the distance or depth in the third dimension, but actually for this we adjust dimension 2 (frequency spectrum). Vocals for instance should sound in front with their trebles active, so here we do not roll off. Choirs can be sent to the backstage with lesser trebles, so here we roll off more. For events at the front select rich reverbs. For events to the back select duller reverbs. If needed use an EQ in front or behind the reverb to set the distance or correct the reverb signal. Don't contradict dimension 2 and 3, setting up ambience reverb for close upfront fundamental instruments and do not roll off the high frequency range to keep it all upfront. Think an avoid contradicting 3d spatial information, use a stage plan and act accordingly. Again the Reverb is placed as a send effect or group effect. In this way we can make use of the reverb for one or more instruments together (group tracks). Specially placed on Group Tracks it can give more welding and layering, togetherness. As a send effect the reverb will not affect the dry signal, thus confirms with our natural hearing. The dry signal is crucial to hearing and must be kept (leaving transients intact). On top of the dry signal is the reverbed signal, so our hearing accepts the distance / depth. The dry signal is always present in natural reverberation hearing. As a creative aspect we could use only the reverb signals, but for perceiving depth naturally we need the dry transient signal to be present as well as the reverb signal. To set the reverb as an insert effect is not common and mostly done out of artistic freedom, still then set the reverb post fader and 100% wet, adjust the reverb controls until sound is correct. Sometimes only one instrument needs a reverb especially for itself, so we could insert a reverb and mix the reverb on the instrument track (for instance the snare). Still routing to a Group Track is best, even if this means just one single instrument is routed to this group. Group tracks or Send Tracks are good for reverbs because they can save processing speed and layer or weld the group for more togetherness, summing up towards the master bus fader. For instance a reverb or delay could do some furthermore welding and blending on a group track forming a layer. Each layer could have its own reverb. First resort to EQ and Compression for Groups. Maybe some gating or limiting. Then route to a reverb (delay, echo, effect). Maybe roll off some lows and highs first. How many reverbs you need inside the mix depends on your mixing technique. But generally four or more reverbs on a basic mix are quite common. A good chosen reverb can re-place other badly chosen reverbs. Sometimes there is little need for reverb and the style of music played needs to be dry (just some ambience), sometimes there is room for a lot of reverb and is needed for creating the space (distance, depth). If required you can add a delay after the reverb, by this way you can spread the reverb signal more (stereo delay, watch the correlation meter) and maybe then becomes clearer to transmit the coherent 3d spatial information and avoid masking. Sometimes just timed events of automation is needed to temporarily avoid masking. When the reverb sits behind the mix we call this the Masking Effect, the reverb is masked by the dry signal of the mix. Individual instruments or tracks. Just adding some delay, a bit of panning can help the reverb jump out of its masking partner and be freed again. As a drastic measurement tool you could use some widening or stereo expander (correlation). Automation becomes handy when only a part in the timeline is masked. If reverb could be synced to tempo this would be worth it on longer reverbs or delays.

Gated Reverb - A setting where the reverb stays at one level over time and then suddenly shuts off. Often heard in snare drum sounds in the 80’s. Gated reverbs are good for keeping rhythmical content. Basically a gated reverb is two devices in one. A reverb and a gate. Reverb 70's effect, set reverb to pre fader, lower the original sound level fader, only the reverb signal will stay.

Decay Settings: Choosing the most appropriate reverb treatment for a song can be surprisingly difficult, especially if you have hundreds of presets to choose from. So, instead of regarding reverb like the glue that holds the mix together, try adjusting its parameters (and in particular the decay time) while listening to the reverb return by itself. If the decay time is too long you'll hear a continuous mush of sound; if it's too short you'll scarcely hear it unless its level is turned right up. Somewhere in the middle you should find a setting that adds rhythmic interest to your song, without overpowering it, making the reverb work for its keep. This is also a useful technique when using several reverbs in a song, to make sure they complement each other.

Pre-delay: No pre-delay? No problem! Some reverb plug-ins, from freeware favourites to tasty convolution types, don't offer pre-delay — a user-configurable gap before the onset of a reverb's early reflections and tail. It's useful to have, though, as it can contribute to the clarity and separation of individual voices and instruments in a mix when large amounts of reverb are used. Using most software DAWs it's straightforward to rig up a pre-delay for a reverb (or any other effect) that doesn't have one. All you do is set up your reverb on an aux track or channel, but place a simple delay plug-in in a slot above it. Set both plug-ins' wet/dry mix parameters to 100 percent wet, and feed them some audio using an aux send on your normal audio tracks. Now the delay plug-in operates as a pre-delay for the reverb: easy! This kind of 'modular' pre-delay actually opens up some interesting possibilities. By using a multi-tap delay, or a simple delay with some feedback, your dry signal can be fed to the reverb several times, making for longer, more complex — or plain weird — reverb tails.

Wet Set: If you have a sound that you want to push a long way back in the mix, it can often be better to make your reverb effect pre-fader, and temporarily remove all the dry sound. Then alter the sound's EQ and reverb settings while listening only to the wet reverb sound. Once you've got that sounding good, gradually fade the dry sound back in until you're happy with the wet/dry balance. This approach can often be more effective than simply whacking up the reverb level while you listen to the whole song.

Combining Reverbs: You don't have to generate all of the reverb sound from a single plug-in, and using two different reverbs can also help you to save CPU power. For example, though a nice convolution reverb gives a good, believable sound, long impulse responses tend to eat up CPU. By using the convolution reverb for the early reflections, and then using something like Logic 's Platinumverb or Waves Trueverb to add the reverb tail — which is less critical to our perception of the sound — you should get a convincing but less processor-intensive result.

Group and Send FX using Reverb or Delay.

To make some kind of combination and the fact that on digital systems for processing speed, we cannot just throw in a lot of reverbs and hope for the best. Most likely your digital system can cope only with a few good reverbs in place. In older days recordings where done in rooms separating players and with multiple microphones to capture dry and reverberation signals. Off course it would be great to use a reverb for every instrument but we can't. Anyway it would be a bit complex keeping track what you needed the reverb for in the first place. For dimension three, we need the reverberation. Therefore we must know why we use the reverb. Commonly used for dimension placement and stage planning. Keeping track means some kind of bargain on complexity and amount of reverbs. Anyway more reverbs mean more mud and fuzz, so keeping only a few good reverbs is the way to go. Maybe 4 to 6 reverbs for a full mix to shine is quite good goal.

Delay after the Reverb.

A delay can help avoid masking of the reverb and can help to be added after the reverb signal to make the 3d spatial information more clear. Only do this when masking does not go away. Sometimes a delay can be placed in front of the reverb. Automation can help parts to unmasked events.

Compressing Reverb And Delay

Using a compressor on a reverb bus can really tighten up the mix, if the reverb tends to be getting too loud and out of control dynamically. Some heavy compression can sound quite nice but be careful not to over do it and remove the life. The same goes for delay busses, compression can really tame the sound and stop anything from going too out of control. Also, using EQ on a reverb or delay bus is a great tool for removing any potential muddiness that may be happening.

Delay and Reverb Techniques

Panning - Panning reverb into the opposite channel (2 channel or 5.1) can produce a very classic and depthful effect. For example, in a stereo recording, you can pan the source into the left speaker, and pan the reverb of the source into the right speaker. A plug-in can be used

Pre-delay - Pre-delay can help with temporal un-masking. Basically, you must tweak the pre-delay of a reverb every time you use a reverb, so that the reverb signal and the original signal does not overlap in time. Many engineers will not use a reverb without pre-delay unless they create the same pre-delay effect by other means.

The Haas Effect - Delay time set beteen 12-40ms, or multiple delays set between 12-40ms - The Haas Effect can be noticed when you set a delay but the original signal and the delay blend and the delay does not become and echo. The delay is happening so quickly that it the source and the delay sound like one. This is extremely useful in creating depth in a mix.

Depth and Space - Reverb using pre-delay and Delay/Multiple Delays using the Haas Effect can help create depth and space if they are used correctly.

Creating space - Some of the best mixes make the listener feel as if they are moving into different spaces. Using these techniques, you can create the sensation of moving from space to space as songs or parts change.

Reverse delay and reverb - In almost all DAWs, you can reverse a recording (so that it plays backwards) then apply an effect, and reverse it again. When you think of that odd effect on modern horror movie preview voices, then you are thinking of the reverse delay. If you have never experimented with this technique, then you will likely be very happy to add it to your tools.


Masking or the masking effect will hide your reverb behind the dry signal, the 3d spatial information that the reverb (or any other effect) is adding will not be perceived as depth or distance. Masking also occurs when two signals / instruments play in the same frequency range from the same direction. Unmasking is when we correct this and have cleared pathways for instruments signals to shine, saving headroom by a reduced level and still have a good mix. Basically we can maybe hear the reverb somehow; it is masked and therefore hidden behind more louder and sustaining sounds. There are some solutions. The first is questioning the instruments that are sustaining (or the reverbs that are sustaining) and are affecting the transients, if this is not needed maybe a compressor can help to clear up some headroom or reduce the sustaining sound or raise the transient sound (or gating). Second is just raising the level of the reverb (common easy solution). But before our ears will understand the 3d spatial information of the reverb, maybe you will raise too much (creating more fuzz or mud and having less headroom left over). A good quality chosen and clear sounding reverb will solve this problem better. Bad reverbs are causing overblown mixes and lose a lot of headroom, still not be perceived as depth. Preparation in dimension 1 and 2 is crucial before adding dimension 3. Then with a good reverb there is little level necessary for our ears to recognize the 3d spatial information. Dimension 1, 2 and 3 are all needed to perceive depth or the dimensions, and make our ears understand the mix content (stage). Just when reverb is sitting behind an instrument, changing the pan or balance on either the reverb or the instrument of question might do the trick and uncovers the reverb (unmasking), panning it the first choice to grab for, and the level next. Anyway dimension 1 panorama and dimension 2 frequency spectrum are coherent to dimension 3. Maybe you decide to re-place a guitar track and set it more left, and then maybe the reverb (or effects) that are routed for the guitar must be looked after also (more to the right). When you add depth to a mix by adding effects, it is better to have dimension 1 and 2 somehow finished then start with dimension 3. The more completed your mix is towards finished, anything you change will have a cascading effect and requires thinking and maybe re-thinking and more work. Whenever a reverb or delay (or both) is masked by other instruments or just can't be heard enough, try to undo the masking effect by forcing the 3d spatial information onto the listener’s ears. With a good reverb that is in place, you won't have to force too much and avoid to fuzziness and muddiness altogether. A mix can be soon muddy and using EQ to correct this is well accepted, but in the first place the sound of the reverb is of importance and panning/level. So whatever signal you input, you better be sure it is cleaned from unwanted frequencies or material. So remember to sort your reverbs out and know what reverbs you like best, this will give you a head start and will avoid the complexity and saves time and frustration in further mixing. If the result is not mono compatible, try two identical reverb presets from two different devices, panned left and right. Both reverb devices receive opposite send signals so that the left of the panorama is reverbed right and vice versa.


Keeping track of things.

It is good to describe the information on a track why you setup a reverb or delay (or even any other effect) and why its settings are there, why you need it. Take care to describe the 3d spectral dimensional placement (stage plan).  Also you can write down all reverbs that you like and keep track of them for later use while mixing. Software and digital mixers sometimes have digital notepads; keep pen and paper at reach. Modern DAW's often provide notepads for a song, track, instrument, mixer, so keep notes and keep track of your info. You might have forgotten the next day, what brilliant solution you had the day before.

Starting a Mix and progression towards a Static Mix, Workflow of a mix.

Until now we have explained how any mix can be started, after recording is done. For a quick overview here is a 'to do' list. Each time we refer to an instrument or track, you can find specific information about this instrument or track below this mixing section. Each instrument referred over here, can be found below. For panorama, frequency spectrum, quality, reduction, compression, reverberation and other specific tools, refer to each instrument's section for details.

0. Recording instruments or tracks must be done in quality before mixing, with quality equipment. Keep the signals noise free by itself, free of humming sounds or continues sounds, do not use a noise reduction plugin or system when recording. Some like to record with the Dolby button on. Try to be careful with placing effects, EQ or compression on recordings in progress. Try to separate and record as much dry. Record in stereo, on digital systems use 32 Bit Float for internal processing purposes. Convert samples / files to 32 bit floating point.

1. When starting a mix, set all Faders at 0 dB, set pan or balance in the middle (center, unity). Remove any EQ or Compression, Effects or Plugins. Set all equipment you are using for mixing to zero, dry, bypass, unity, center, etc. Re-set all on your mixer to the most basic starting position.

2. Sort out your tracks from left to right on your mixer. Placing more fundamental instruments or tracks at the left side. Spreading to the right side. Label every track. The tracks from left to right could look like this, by example, Basedrum, Snare, Claps, HI hat, Overhead, Toms, Crash, Others, Bass, Guitar 1, Guitar 2, Piano, Epiano, Keyboards, Synths, Others, Main Vocals and Background Vocals. Next to the Main Vocals on the far right there is place for each send track then summing all up towards the master bus fader and output. This can be debated, but you are free to setup your mixer anyhow you like. Modern small mixers or controllers have only place for 8 tracks at a time, spreading drums on channels 1 to 8, and the rest on channels 9 to 16, can help for switching back and forth on the mix setup (especially when using mix controllers). Label sort and color code tracks, assign them to group tracks, folders and route them. Use the group solo function to control the routing. Prepare the mixer for a new start (starter mix).

3. Listen trough every track (in solo mode), cutting out any unwanted signals like noise, pops, clicks or rumble. Any unwanted material must be removed, either audio samples or midi, first choose to do this on a manual level (manual editing). This is tedious and some really like this fase or dislike. Some remove breathing noises manually from vocals or some use a de-esser. This manual editing may seem as a bit time consuming, better to remove and be sure you are only hearing what you need. When you are using a sampler, you could clean the samples before using them. How you do this is not of importance, but take some time to clear up and clean up. Once you start mixing and listen to your whole mix or combination of instruments and tracks, unwanted sounds might get hidden inside the mix (masking) and are uneasy to find their location. So clean up while you can, when you can. Check each track, for breathers, editing mistakes, clicks, clean them. Only reduce noise or humming when needed, better to have every recording clean before using any noise reduction system.

4. Define your mixing strategy with a panorama sketch. Draw a sketch of a stage plan and place all fundamental instruments Basedrum, Snare, Bass and Main Vocals first, then the rest of the not fundamental instruments alike the rest of the drum set, guitars, organs, pianos, keyboards, strings, percussion, background vocals, use panning to keep them out of the center, decide the frequency spectrum and depth. Try to be natural and consistent, separation as well as togetherness, use counterweight or counteract.

5. Mute. Mute all folders and tracks, with exception of the drum folder. Start building up the rhythmic backbone, starting with the bass drum, followed by snare, making use of panning, EQ ing, Compression, gates and reverb (delay) until the drums represent a powerful and rounded sound. If the bass is part of the drum group, add it to the mix after editing as required. Your next step is to build up the instruments that provide harmony and warmth. Distribute them according to their complimentary spectral properties to the left or right in the panorama. Create a good lead vocal sound and add it to the center. Balance the group levels of all groups edited so far. Distribute decorations and additions in a spectrally sensible manner around the existing basis. If an event sounds fuzzy, look for a spot within the three dimensions where it can be heard. If you cannot find the spot either with a good panning strategy or with EQ ing or layering, reconsider the reason for having this event at this place in time. Fine tuning of volumes at extremely loud or quit levels.

Do a first check on the whole mix. Set the master fader at 0 dB. Set all balance to center and adjust all faders until you’re a bit satisfied. Do not use EQ, compression or effects. You’re looking for mix that is quite straightforward and that comes from one direction, all from center. By only adjusting each fader until you find some dry mix that works for you. This must be easy to setup and only takes a few moments to do so. Don't worry and fiddle to much, we can be more precise later on. Also you could use some EQ to sort out the bottom end of your mix. Using a low cut from 0 Hz to 30 Hz for Basedrum and Bass. Using a low cut from 0 Hz to 120 Hz (180 Hz) on all other tracks or instruments (including the rest of the drum set). Adjust the starting frequency of the cut, just below the last main frequency. Keeping what is needed and deleting what is not needed, just use some EQ for adjusting the bottom end. At least by doing this we guarantee that Base drum and Bass have a clear path and the mix is cleared from any rumble, pops or click in the bottom end range, as a result we now have more headroom. For some distance and reduction the Base drum and Bass can be rolled off in higher trebles. Setting some distance on other instruments or tracks according to our stage plan, roll off some more highs. Do not pan the mix for now (keep dimension 1 and 3 unaffected). Just apply some reduction, quality, headroom, separation and togetherness.

Listen to your dry mix for a while. Decide by experience how to plan the dimensions. Draw a quick picture; plan the stage inside the three dimensions. Plan the fundamentals alike Basedrum, Snare, Bass and Main Vocals in center. And build the rest of instruments around this (not fundamentals), placing not fundamentals more to left or right, don't be afraid to pan. Anyway do not touch anything right now, just think of it or draw a quick sketch on paper. First we set dimension 1, panorama. Pan first before setting fader level again, apply the panning law and know that relative volume of a signal changes when it is panned. Completely apply all panning first, then adjust all fader settings until are satisfied. Keep adjusting balance (panning) and fader (level) until satisfied with your stage planning for dimension 1 (panorama). Listen to your dry panned mix for a while. Fader and Pan are most important settings to start a mix and mostly overlooked, so we tend to take more time here for listing and adjusting.

6. By having some notion now where to place instruments, it is time to listen and decide what instruments need EQ or compression in order to adjust its frequency range (dimension 2) and coherence to other instruments. Also by doing this we can save some headroom. We can try to adjust for quality and reduction. We have made a separate instrument section below for reference. Anyway we need to adjust every instrument for its spectral content, mostly doing EQ where needed. A steep filter for bottom end cut offs reduction (separation, saving headroom). Headroom in the bottom end (0 - 120 Hz) should be only for Bass and some Lower Basedrum thumb sound (kick), cutting all other instruments in the lower range. For the whole mix and for the Bass and Basedrum (or any other fundamental instrument) is what we are after mainly. For quality each instrument or track can be adjusted until sounding good, keep in mind not to fill the misery area from 120 Hz to 350 Hz. Try to avoid boosting the Mids of all instruments, instead choose a few, leave the rest or cut. Mainly using tools alike EQ, Compression, Gate or any other dynamic tool. Also for distance we could roll off some highs for each instrument or track. Remember when you adjust an instrument or track, to bring the level back into the mix directly afterwards.

7. Now solo the most fundamental instrument (likely it is the Basedrum, but however some start with the main vocals). The Basedrum should be on the left of the mixer side. For example we have chosen the Basedrum as most fundamental instrument here and is most common to do so. Solo play the bass drum and watch the master vu-meter. Keep the level at -6 dB to -10 dB on the vu-meter by setting the base drum fader level accordingly. Next add the Snare or Bass (you decide) and use its fader to set the level. Do not touch the base drum fader; adjust the instrument level you’re working on only. Each time you add a track to your mix, set the corresponding fader level. Until you have worked your away trough the right. Looking for some togetherness and having levels just right for a dry mix is crucial over here. Just keep adding and adjusting until finished. When finished then do a check on the vu-meter level, you must have some headroom left, else set all instrument or track fader back the same amount. Leaving some headroom for later on.

8. Decide how you are going to separate the Basedrum from the Bass. Making them sound well in the lower frequency range. Start off with the Basedrum (listening in solo mode), roll of some subs from 0 Hz to 30 Hz (50 Hz), roll off some of the highs > 8 KHz for some distance according to stage planning (behind main vocals and bass). Creating a good Basedrum sound. For quality and reduction on the Basedrum refer to the instrument section below. Maybe add just a tiny touch of reverb, with little pre-delay (no pre-delay actually for rhythmic content). Only use an ambient reverb or small room/drum booth reverb, we can use for the whole drum set so can be on a group or send. Then aim for a nice -6 dB to -10 dB level at the master Vu-Meter while playing. Remember this is your reference track or most fundamental instrument. This reference fundamental instrument is used to set all other instruments after. Instead of the Basedrum, maybe the Main Vocals or any other instrument could serve as most fundamental. But keep in mind that likely fundamentals are lower frequency instruments or tracks, as we need the center of the speakers to produce the lower bottom end fundamental frequencies (left and right playing together). Accordingly and measured off by this reference (most likely the Basedrum track), keeping always in the center of the panorama. Also it is best not to sway around in center, just keep it dead center or the added signals must refer to center. Left to right time lined events are not recommended at all. Keep your most fundamental (bass drum) instrument in center at all time. So listen through the whole bass drum track solo, adjust it, and just be certain it stays always in center all the time.

9. Next, For the Bass just roll off some very low subs (0 - 30 Hz) and roll of some highs ( > 8 KHz). Solo the bass and create a good sound, refer to the bass instrument section for this instrument specific while mixing. Listen to both Basedrum and Bass together, then only set the Bass fader and listen to the combination of Base drum and Bass (do not touch the Basedrum fader, this is your static reference). Do anything (add EQ, Compression, etc) to correct the bass signal now. Set the level of the Bass until it feels and hears correctly (togetherness). Also keeping the Bass in center always.

10. Then introduce the Snare. For this you can decide to solo the Snare, apply a low cut for separation and create a good sound (see the snare instrument section). Snare usually needs a larger reverb. Do anything you need to correct and enhance the snare signal now. Then in combination with the Basedrum set the Snare fader (solo Basedrum and snare). Introduce the Bass and keep setting the Snare fader furthermore. Maybe you do not find right settings at start, keep fiddling soloing and playing together. Setting only the Snare or Bass faders. Just find some fader settings that workout best, then leave them alone.

11. Introduce the Main Vocals, first in solo mode. They must be upfront, so no trebles cut over here. Just roll off the bottom end to separate from Basedrum and bass. You can always adjust the fine roll off frequency later on, when you unhappy with the vocal sound. Use a stereo EQ filter setting to balance the vocals in center even more. Then try to make a good sounding vocal (see the main vocal section below). This can mean dropping a de-esser in place or delay / ambience room reverb (we already have one in place for the drums and bass. Or use some fine EQ ing to get the vocals really sound correct. Maybe some compression. Then un solo the vocals and again adjust its fader to set the mix. Remember vocals must be heard clearly upfront, when not reconsider now.

12. Then add the HI hat, placing it with balance slightly right, according to position. Roll of a great deal off the lows from the HI hat. See the specific HI hat section below for more details. Add the overheads and give them some distance by rolling of some highs. Maybe a Stereo Expander can widen the overheads, watch the correlation meter for mono compatibility issues.

13. Continue to add each drum set instrument until finished. Adjusting only the newly introduced and stay away from earlier introduced instruments. Workout a good steady sound for drums, spend some time to create and finish off the drums first. Drums are important, they sound so much better inside a mix when first completed as a drum set. Only continue when happy and completely finished all drum set events / instruments.

14. Add Guitars, Keyboards, Synth, Percussion and any other instruments or tracks. Remember when you place something left or right, you need coherence, so counteract. We need to counteract instruments, we can counteract instruments and their reverb signals. Keep away from the center and be creative placing them left or right (be courageous). So placing a guitar left might need the keyboards placed right as an opposite coherence (counterweight). Work out your mix in dimension 1 - panorama, pan, balance and fader level first. Then adjust dimension 2 - frequency spectrum of each instrument by adding EQ or compression as an insert effect on each individual instrument. Cut lows and highs where needed. Also EQ and compression can adjust the internal fundamental frequency range, so making your individual instruments sounding best is off course recommended. According to our stage planning, we try to stay within the dimension 1 and 2 boundaries more and tend to place dimension 3 later on.

15. When you have some choirs you do need them into the back of the stage, so roll of some highs and for keeping them out of the lower frequency range roll off the lows also. Also here on the choir we can use a stereo expander to widen the choir in the background. According to panning laws we spread the background vocals or choirs (lower voices more centered and high voices more outwards). By widening the overheads and choir we keep them away of the already crowded center path.

16. Next, for the rest of the instruments (all instruments) decide where to roll off more bottom end in order to keep the lower frequency range of your mix only available for Basedrum and Bass, to separate and avoid masking, leaving some headroom. Also avoid instruments not needed to play inside the 120 - 350 Hz misery range. Solo first, cut where needed and create a good sound (use your stage plan and the dimensions). This can mean some heavy balancing, EQ ing with some steep filters or compression. Just not to interfere. Repeat until you have finished off all instruments. Remember it is not recommended to adjust an instrument or tracks fader after it has been set. Do anything to make the track / instrument / sound better now. When working each instrument or track, do try to adjust that track only, without adjusting the other tracks.

17. According to your stage plan, you must now have setup Level, Balance and Frequency Range for each separated instrument or track. And maybe already have rolled off some high trebles on some of the instruments or tracks that are more distanced, all according to our stage plan. We placed only dimension 3 when needed, mostly ambience for upfront instruments and larger duller reverbs for more distanced instruments.

18. Listen to the Drum set, Snare and Bass together. Maybe create a Group Track and route them to it. This will be your first Group of many to come. Some do like to route the drums to its own group, therefore you can also route the bass to its own group. This will keep them separated as an individual instrument groups.

19. Next assign groups to instruments that are close to each other and can form a layer together. Maybe a group for guitars. Another one for piano, Epiano and keyboards (synths). A group for the background vocals (choirs) and a group for the main vocals. Assign Group Tracks for each range of instruments. For now do not use any effects on the groups. If you like to use an enhancer while mixing, use it on a separate group and route only instruments or tracks that need to be upfront, but we don’t use it for now.

20 Try listening to the whole mix again, by muting or soloing instruments you can find out if the placement of each instrument or track is correct and according to our stage plan. Else keep correction dimension 1, 2 and 3. Be sure you have found some kind of clean sounding mix before you go on that is exactly according to our stage plan. If not keep fiddling about until you are satisfied. Try to stay inside dimensions 1 and 2 by using only Fader, Balance, EQ or Compression (gate, limiter). Then correct dimension 3. Maybe this will take an hour or so, it is crucial to get it correct.

21. Listen to the whole mix and decide it's level, pan, balance, EQ and compression, gate and limiter are setup correct. If not keep adjusting the mix until satisfied. Working only instrument or track based (see the specific instrumental details below). We tend not to use any effects on groups, sends or on the master track.

22. Now we should have a mix that is clear (dry), where instruments can be heard, still have some togetherness and have some idea of the dimensions 1,2 and 3, sounding correct as planned. Even with separation that is contradictive to its opposite togetherness, it is possible to have a combination of both. A mix thrives on separation from start (dimension 1 and 2) and get some kind of layering. Only adding some reverb or delay in dimension 3 to create some depth, when we are sure that we are happy with dimension 1 and 2 first. Be aware of masking and learn to understand it very well, learn how to unmask.

23. Now it’s time to glue the mix more by adding to the groups where needed, hopefully we have created enough headroom for mix adding purposes. EQ and Compression on a group can weld or glue instruments together. Making groups appear as layers for mixing purposes. Summing up towards the master bus fader output. Compression on a group can give a feeling of a layer (togetherness, glue or welding) and give some coherence of grouped instruments. Also by using EQ in front of the compressor (only place an EQ or compressor when needed) you can sort out the frequency range by cutting lows or highs (or compress), by this way the threshold of the compressor will only react to a cleaned dry input sound. You must cut lows when they are not needed, affecting other instruments in their range that are more fundamental. You can cut highs (trebles) when you need the group to be set back into distance (depth) or when you know these frequencies simply do not exist at all (preventing noise, humming, clicks, etc). Planning the three dimensions finally. Remember that Panorama is first looked at and adjusted, then Frequency Range, then Depth, follow the dimensions.

24. For working out depth on a dry sounding mix we can use reverb or a delay (most common) to give some space (dimension 3). The group tracks are likely places to add 3d spatial information (placement, depth) to a mix. So a good reverb or reverberation effect on a group or send track will give room characteristics and placement. As we did combine instrument sets, we can now use the groups function to combine overall effects (alike reverb, delay, compression and EQ, etc). Somehow you know you need at least a few reverbs for Drums, Snare, Bass and Vocal alone (ambient small room or drum booth), we can maybe route al instruments needed to this group. Each group can differ in room and reverberation settings (see the specific instrument details below). Place a reverb where you need it mostly, but be scarse. Rather on groups. You can decide to place a reverb on its own single track or on a group, depending on the purpose. By using groups (or sends) you can save just some reverbs though and keep it tidy. Now you understand you have at least a few good reverbs running in the mix, just to sort out dimension 3. Choose good quality reverbs. We did not even consider using reverb as an artistic (creative) factor; reverb or reverberation is common for dimensional placement (3d spatial information). You can understand that a mix with 4 to 8 Reverbs is common, because almost each different set of instruments (Tracks, Groups, Sends, Layers) need placement and depth, as well as some welding for togetherness. Use a bright reverb for upfront instruments or tracks, and a duller reverb for distanced instruments or tracks. Use compression on a group when you need to weld them more together. Summing up groups towards the master bus fader output.

25. With all these reverbs in place, avoid muddiness. So maybe EQ or filter the lower bottom end of the reverb with a good cut from 0 Hz to 30 Hz (50 Hz or much higher) on fundamentals. And a good cut from 0 Hz to 120 Hz (at least, > 180 Hz) on not fundamentals. When instruments or effects are masking separate them with Balance, Pan, EQ, Compressor or some delay after the reverb. Or in more extreme cases use a stereo expander after the reverb to even make the panorama wider, maybe use timed events as automation (just when we need it as a last resort). Crowded mixes could be widened and listened as if played outside the speakers; this gives some more room in the field (stage planning). Basically be reluctant to place the stereo expander and use only as a last resort. Watch the correlation meter, goniometer, when you are using the stereo expander or are working in dimension 3 with reverb. Check for mono compatibility, maybe you can leave the correlation meter in visual sight always. According to your dimensional planning now add depth in dimension 3, by adding those reverbs (delays) that are really needed to create depth and transmitting the 3d spatial information to the listener. According to our stage planning, some instruments need to be upfront and some more set backwards. If a set of instruments alike a drum group needs a particular reverb, place the reverb on the group track. If a reverb only affects the single track or instrument (snare for instance), place the reverb on the single track (snare track) or even still place it on a group so other instruments can make use / benefit. For instance with the snare reverb, you can place it on the instrument track to make a difference, but this will keep it from being used by other instruments. Try to have reverbs and delays available on group or send tracks, instead of using them on single tracks. Transients are always fist recognized by human hearing for calculating depth and distance; we need the dry signal to be present (transients, on top the reverb signal). The dry transients must be heard, as well as the reverb signal (also the transients from the reverb signal as well as the original dry signal). So our mixing tactics must include all necessary transients to be heard, to be perceived as natural depth (dimension 3). Any confusion created by not applying dimension 3 correctly will affect listening pleasure. Any conflicting information will confuse the listener.

26. Now here is where the routine starts to fade. As you have setup a mix that is consistent in placement in the three dimensions. Work around the mix for some togetherness and clarity balance (separation, quality and reduction), now it’s time to be more creative and invest some time. But with following guidelines 0 to 25 we at least have started a mix with some rules or routine. Anyway to finish off what you have started, do a check, a re-check and a double check on your placement. Check levels, peaks, frequencies. Use hearing, listening and visual methods (spectrum analyzers, correlation meter, goniometer, peak RMS meter, etc) to come to the right kind of decisions or conclusions. Now you should have a well sounding static mix where you can add more quality and effects or automation, because you have some headroom still left inside you can be creative and mix further towards the end result.

27. You could place a Limiter on the master track, just to avoid some peaks. But on most digital systems you will be signaled by a LED light when you’re passing over 0 dB. Anyway even on 32 Bit Float digital systems, going over 0 dB is not recommended. On a 24 Bit or 16 Bit system always stay below 0 dB. When you do use samples or audio repeated times inside your mix, from drum samplers or instrumental samplers, it would be a hassle to find out what bit rate they are all played in. Better use a common bit rate and sample rate (preferably 32 bit floating point). So staying below 0 dB everywhere seems to be a good solution of not harming any parts of your mix (else convert). If you have a limiter placed on the master track, just be aware it's for peak scraping only, mostly a Brickwall limiter with a threshold of -0.3 dB or a low peak reduction of 1 dB or 2 dB. Do not tend to use any more limiters. When your mix tends to be too loud and attacking the master limiter, then set back each instrumental fader or their corresponding group fader by the same amount (creating some more headroom). Do not touch the master fader; this will always stay at 0 dB. Only when your master fader is the last control to your speakers as an amplifier you can change it scarcely, better is to find a solution to keep the master fader at 0 dB at all cost. Listen you mix at loud and soft levels. Sometimes instruments disappear when playing at soft levels. A mix must stand loud and soft. But most of the time listen to soft levels while mixing, do not over excite your monitor speakers as well as your fatigue ears. You can train your ears better working at softer monitor levels.

28. Summing up - When you understand that Bass and Base drum should occupy the lower frequency range of 30 Hz to 120 Hz only, without interference of other instruments. Bass and Bass drum should own the lower range by themselves. This can mean all other instruments and its effects are somehow or completely cut in the 0 Hz to 120 Hz (180 Hz) frequency range, thus avoiding the bass range. Spend some time working on the misery area from 120 Hz to 350 Hz, where most instruments have a piece.

29. Keeping other instruments (not fundamentals) in their range and place them left and right (opposite to each other’s opponent instrument to counteract) is keeping them out of center. Center is the place for Basedrum, Snare, Bass, Main Vocals. Keeping the main vocals upfront by not cutting off its trebles. For setting other instruments more left or right (away from the center path), does not mean they are perceived as such. When they are accompanied by 3D spatial information (pan, frequency, depth) as in form of a reverberation sound opposite to its dry signal, cutting off trebles to make some distance, you are placing them inside the three dimensions. Reverb or delay can work as a counterweight. When placing a dry instrument to the left, maybe a reverb placed on the right can work as on opposite filler. Also comparison instruments can work as opposites. We tend to layer them in groups. The groups finally can be used working towards a finished mix (static reference mix). Using techniques on groups alike EQ, Compression and effects for more welding together. Using groups or send for our reverberation needs. Mostly we do not like anything on the master track, some like a limiter in place to scrape some peaks or for peak warning messages. Try to watch the master track vu-meter while mixing and try to keep it below 0 dB.

30. Keeping a balanced overall sound coming from both speakers is planning the three dimensions and mixing towards this goal. Avoid masking of reverberation by adding a touch more or pan (balance) them away. Sometimes a stereo delay might work behind the reverb to avoid masking. Watch the correlation meter for mono compatibility. Do checks and re-checks and make sure your planning and mixing rules are applied. Listen a mix dry (without those reverbs), check if you have not used too much reverb, but just enough to transmit the 3d spatial information to the human ears.

31. Quality is a general rule. Off course it is important how any separated instrument is sounding in quality. So while busy generating a nice mix, individual sounds (solo) make up the mix (summing). You can adjust any sound, track or instrument anyhow you like, with the use of EQ, Compression and other effects. Beef it up, make it nice. So when using effects (especially reverb) do not hesitate to use the best instead of using the most efficient. Avoid muddiness and fuzziness, apply separation (use the dimensions, the whole stage). Use the rules for quality and reduction. Use panning laws. Refer to the specific instrument details below. And finally as all instruments play as a mix on the master track, your mix must sound dam good! Only when you’re happy with your mix, as is, you should continue. Else revert to the basics of mixing (repeat the mixing steps above), add or remove until you’re happy with your final sound. As a final static mixing stage, adjust the groups or just play around with them until you find a nice coherent static mix.

32. Until now we have worked on the starter mix towards static mix, and have finished it until satisfied. Basic Mixing III will explain dynamic mixing. But for now we will skip the dynamic mixing and jump to a pre-master. A pre-master can be a good tool to hear and analyze the mix, before we continue with the dynamics of the mix. What final sound of a mix is best? Well, this is more complicated to explain, according to style and preference. But maybe you can remember this chart below? You can read about it in Basic Mixing I!

33. Volume automation for introducing events. Volume automation for song structure dynamics. Panorama and stereo expander automation for clearing up the last remaining fuzzy spots. Carry out further automation. Creative fine tuning to refine details. Constantly experiment in order to improve events that do not yet sound right. Set the brick wall limiter in the master section to -0.3 dB. Export mix down, 32 bit, no fade in or out and a bit of clean silence. Use the mute button composition wise when needed or create new pleasant combinations in time. Remember the more instruments play at the same time, the more worries and corrections needed. Also a song or track will sound dull and equal when all instruments plat from start to finish, consider composition wise events and cut when needed. Less is better than more.

Anyway, repeated mixing will give you a beforehand understood notion what a finished static mix should sound alike. Experience and understanding might be the main factor for learning to apply. For checking a mix, a spectral analyzer can be a worthy tool visually. For instance you could check your mix against other commercial recordings, using the A/B method. AAMS Auto Audio Mastering System can be good a tool to help you analyze your mix and get some suggestions for better mixing results. You can train your hearing by listening a lot of good commercial available music on your mixing monitors. Or just listen to a lot of commercial available music anywhere you can. At least you know what quality your monitor speakers will play. And you know what commercial music is sounding alike. When in doubt while mixing, take some distance again. Compare your mix to other music. Fist Revert back to dimension 1, then 2 and then 3. Check how much headroom you have left. Listen with clean ears, listening hours of music can make your ears fatigue. Then it might be good to leave the mix for the next day and start with a fresh mind and fresh ears or just take a good (>15 minutes) nap. Sometimes this is needed to really interpret well. Also pre-mastering a mix can help clarify more. You can use AAMS Auto Audio Mastering for this purpose. A mastered mix is perceived louder and also stands up more against other commercial recordings. Pre-mastering can reveal sometimes more (what is good, what is bad), even when not heard inside the mix, this suddenly becomes clear in the pre-master. Let somebody else listen to your mix (pre-master) and you will get some feedback, depending on the style of your music mix and this persons dislike, choose how to interpret this advice or criticism. Don't be worried by other people their critics, use this to your advantage. Do not bypass or hurry, you will never get anywhere near a finished mix bypassing the rules of engagement, bypassing natural laws of sound. A finished starter mix to static reference mix takes up to 4 hours of time (maybe more or less). A well finished static mix can take up to 12 hours of time. Together finishing off the static mix altogether can take up to 16 Hours. Remember that it takes the dimensions, quality, reduction, separation as well as togetherness to finish off a mix completely (static reference mix). Better be educated about these subjects and purposes, if not you could be stumbling with mud and fuzz for a long time! When you know by experience what you’re doing, time will decrease fast. Only continue with dynamic mixing when satisfied with the static mix!

FX example.

Send FX1, < 600 ms, Small Reverb, Ambience on Drums and some Bass, no or little pre-delay, slight trebles roll off (overhead, bass drum, loop, bass, snare, etc).

Send FX2, 1/4 note delay, medium to large reverb space, snare, no pre-delay, no treble roll off. Shorten the snare track with a gate. Experiment with a thick gated reverb.

Send FX3, > 1200 ms, big room, background events, chorus strings, up to 60ms pre-delay, trebles roll off strong.

Send FX4, 600 - 1200 ms, ambience, lead vocals, no pre-delay or 1/8 th note, no roll off trebles.

Send FX5, Decay depends on style, delay or reverb delay combination, lead vocals if needed.

Send FX6, Decay depends on style, guitar & keyboard if needed, L10/R20.

Send FX7, Delay effect, strong instruments (vocals), solos.

Send FX8, Chorus.

Send FX, Reverb layering. For instance give percussion tracks a medium, thick room with quality. The return is processed with a little widening to counteract the masking effect and to place the percussion behind the drums. A little pre-delay on the reverb and slightly attenuated trebles.

The frequency spectrum of a mix.

Frequency Range 0 – 30 Hz, Sub Bass, Remove.
Frequency Range 30 – 120 Hz, Bass Range, Bass and Basedrum.
Frequency Range 120 – 350 Hz, Lower Mid-Range, Warmth, Misery Area.
Frequency Range 350 – 2 KHz Hz, Mid-Range, Nasal.
Frequency Range 2 KHz – 8 KHz, Upper Mid-Range, Speech, Vocals.
Frequency Range 8 KHz – 12 KHz, High Range, Trebles.
Frequency Range 12 KHz – 22 KHz, Upper Trebles, Air.

Instrument Rages.

Frequency range 30 Hz - 120 Hz, Kick and Bass, Bass Range. 
Frequency range 120 Hz - 8 KHz , All instruments.
Frequency range 8 KHz - 22 KHz , Cymbals, Hi percussion, High range of all instruments, Air.

Between, 1 KHz to 2 KHz. Irritating. Perceived as loudness for beginners.
Between, 350 Hz to 1 KHz, Generally it can be worthwhile applying a cut to some of the instruments in the mix to bring more clarity to the bass within the overall mix.
Between 350 to 2000 Hz , nasal, woody and piercing. A mix can sound nasal over here.
Between, 2 KHz to 3 KHz, Generally often used to make instruments stand out in a mix.
Between 2 KHz and 8 KHz, is speech related, vocals can shine over here. 
Between, 6 KHz to 10 KHz, Boost, to add definition to the sound of instruments, edge, and ring.
Between 8 KHz to 12 KHz, Treble range, cymbals, high percussion, s-sounds, chimes, etc.
Between, 10 KHz to 22 KHz, Trebles area, follow stage plan for setting distance (Roll Off).
Between, 12 and 22 KHz, Upper Trebles are air, can aid a mix but overdoing is worse.

General and Specific Instrument Details.

First let's explore some instruments and basic settings for the whole frequency range of the mix. It is not important when an instrument sounds awful in single mode, it is important when it sounds goof in the mix. Deeper not fundamental (sure even fundamental) sounds spread in circular form and can hardly be detected below 100 Hz so avoid, whereas high frequencies spread directionally and are easy to detect. First of all the panning rule for fundamental instruments is in center, not fundamental instruments are not centered but placed more outwards.

Between, 0 Hz to 30 Hz (50 Hz), Bottom End. Stay away from the bottom end range unless you are mixing with and for sub. It allows to get rid of all sub-bass artifacts 100% with a very steep downwards cut. Mostly this range from 0 Hz to 30 Hz (50 Hz) is heavily reduced for all instruments, tracks, effects or sounding events, fundamental or un-fundamental (cut). The bass takes the lowest one-and half octaves in the center, not a place for any other instrument, keep free for bass. Above that is the bottom sector of the bass drum 80 to 100 Hz, small banded the thumb or kick. Between, 30 Hz to 120 Hz, Bass range. This bass range frequency is mainly for Base drum and Bass only. The only instrument that can go as low as 30 Hz is the Bass, therefore the Base drum could be cut from 0 Hz to about 60 Hz. Do carefully remove all other unwanted instruments, tracks or events happening inside this bass range (0 Hz to 120 Hz). Cut all other instruments, tracks, events heavily with a steep cutoff filter, be wary of boosting the bass range frequencies. You should be able to find the instruments main lowest frequency and cut just below this frequency.

Between, 120 Hz to 350 Hz, Misery Area. A frequency range where most instruments play, inside this misery area range, almost every instrument will play some of its main frequencies. Best left alone for the most part. But however you can make an outstanding mix when you know to work inside this misery area range.

EQ Tricks. Use a shelf filter type at 300 Hz and sweep around, then you will hear the instrument more distanced or not.

Panning tricks. Use Stereo EQ, work stereo, so we can frequency spectrum pan the whole signal, more left or right sounding. Or use a Dual Panner.

Reverb. To place an instrument using reverb we can use different techniques for planning our stage. The further as sound, the less high frequencies we hear. To distance the instrument (further away): Reduce Volume. Cutoff high frequencies. Cutoff even more highs on background sounds. Make early reflections sound louder than the dry sound. Have a long reverb tail. Do not use the enhancer. To approach the instrument (closer): Use enhancer, lifting area high.  Reverberation must be panned very wide. Make the reverb sound bright, short and dry. Use a short delay and pan wide.


Instrument Effects.

Cut high frequencies of every synth instrument or effects, (12 KHz - 20 KHz).

Drums in General.

Panorama: The location of the drum set is crucial, according to your stage planning, try to keep it natural. Basically Base drum and Snare are panned at center, the rest of the drum set more left or right (according to drum set position, keeping them out of center).

Quality: Drums need to be at a constant volume. Rarely do they change in level throughout a tracks or mix.

Reduction: Apply a good steep low cut from 0 Hz to 30 Hz (50 Hz) for the Basedrum. Other instruments of the drum set can be cut from 0 Hz to 120 Hz at least, to keep the bass range clear. For every instrument inside the drum set, roll off some highs anywhere from 10 KHz to 22 KHz to set the distance (must be behind Bass and Main Vocals according to your stage plan). A frequency component between 0 and 1Hz is called DC offset and must be eliminated, use a the DC removal tool for this purpose. The misery area between 120 and 350 Hz is the second pillar for the warmth in a song after 0-120 Hz, but potential to be unpleasant when distributed unevenly (L C R, panning laws). You should pay attention to these range, because almost all instruments will be present over here on a dynamic level. Cut all frequencies lower than 100 Hz - 150 Hz from all instruments except bass and bass drum, use elliptic 6 extreme cut.

Compression: Drum compression is an art to say the least. The different amounts and styles of compression can completely and utterly change the way the drums sound. Knowing how to compress can save you from a weak sounding mix. The attack and how big you make the transient peak is the most identifiable part of a hit. A too fast attack setting will cut the transient peak and your drum won't hit hard. But if you have a slower attack that initiates the compressor right after the transient peak, it will accentuate the hit (transients). The compression after initiating and bringing down a part of the sustain, the signal falls below the threshold and slowly releases bringing up the decay making the drum last longer and sound larger and more full. This is really a very general overview; you must experiment and listen to find the desired sound. Percussive elements (drums) with long attack time settings (10 ms to 30 ms >) enhance the transients, some more assertiveness, punch or bite is applied to the transients this way. Also setting up the compressor in Opto mode, allows percussive instruments to behave faster and that is a good thing. For all drums that are directly rhythmic and percussive use Opto mode. This will get your drum set more clear and defined. Keep the Snare, Bass drum and some HI hats short (only the transients pass the compressor unaffected) while reducing their sustain. You can always recreate a bit of depth by introducing ambience with a good quality reverb. By using a ducking gate or side chain compressor, you could compress the rest of the mix or certain instruments instead. Bass with Basedrum compression on a side chain.

Reverb: Drum rooms or drum booths (ambient) are recording industries standard on drums. In the early days when only acoustic drums where around, the only way to separate the drums from interfering with other instruments was placing them in separate rooms. So the microphones used could only pickup what was needed. We perceive drum booths as natural (ambient). Drums are mostly placed back on the stage (behind vocals and bass) cutting out some trebles from the reverb signal to create depth or distance. You can give drums some reverb but not too much, just enough to transfer the 3d spatial information, be scarce, only the snare needs a larger reverb or more ambience. Be scarse to use reverb on drums. Only when you mute the reverb, you will notice the change to the dry signal (inside the whole mix, do not solo). Only give enough reverb for drums to transmit the 3d spatial information to the listener. Listen dry and with reverb and decide how much is needed. In the drum section, apply a little less reverb to the Basedrum then to the other drum tracks. The Basedrum can sound flabby when too much reverb is applied. The snare can have a larger and louder reverb. Mostly the reverb tail will end rhythmically short, just before the next beat or bar appears. Reverb can be long (tough interfering with the rhythmic content) and afterwards shortened with a gate, to make them rhythmically stand inside the mix we sync them to tempo when we can. Avoid mud by setting a low cut EQ in front of the reverb or behind (cutting). When reverb is applied on drums, try to make the reverb sound in rhythm with the dry-signal. Maybe by gating the reverb sustain in sync with tempo. Use mostly no pre-delay or < 10ms, checking the rhythmic (we can use a high treble’s roll off for setting distance instead). Missing spatial information can make drums dry and unnatural, so enforce enough reverb so that the depth and distance is clear to be heard, use the best reverb you can find. When reverb becomes obvious, most likely you have gone too far. Just set the reverb so that the 3d spatial information comes across (enforced), but is not overcrowding or too powerful. The more natural the reverb sounds, the better (quality). Try to stay away from the pre-delay, set it at 0 ms to stay in rhythm. Use small rooms or ambient reverb. With a bit of pre-delay from 0 ms to 10 ms (0 ms please for rhythmical content). Just roll off some trebles after the reverb to set the stage plan even more. Try assigning the toms, snare and HI hat a short crisp plate program, with a reverb time of 1.2 seconds. A reverb with a longer decay time can be used on the overheads. Cymbals can be enhanced by a longer reverb. Generally, up tempo songs require a shorter reverb time to allow the reverb to decay between beats and thus avoid blurring the sound (sync).


Sound: For more Kick drum, use sine-wave 60 Hz, Juno 60 Saw tooth. Set pitch for a high note and run this with the original bass drum, this is a house style bass drum. Combine a short transient kick with the HI hat or with a low frequency release from another. Add a closed HI hat (higher click) on the kick transient part, to add click in a mix. Even better to combine two bass drum samples into one, using the first part of the kick from one sample and the sustain from another. Bass drums have two components, kick and sustain highs.

Level: When bass drum and bass sound good, the mix is easier to achieve. Basedrum and bass. The levels between bass drum and bass should be, bass drum is -1 dB or -3 db on the vu-meter. Usually the bass drum will more or less disappear. We can cut some frequency range from the bass, just sweep around 60 - 150 Hz and at one moment the kick will come back (thumb sub kick will return and be ear able again). Also choose the kick and bass its timbres. As another resolvement we could delete all notes from the bass that overlap the kick, mainly 1/2/3/4 measured kick overlapping (start of the rhythmical bass drum content, midi, note deleting or sample cutting). Pay attention to the duration of a kick, it should correspond to the speed of the track. A beginners mistake will be raising the kick at 60 - 120 Hz, this will raise the bass drum but only will be of more level, not consistency.

Panorama: The Basedrum belongs in the center (fundamental). Also in the timeline the Basedrum belongs in dead center of the panorama (specially the lows). If at any time the Basedrum is more left or right, adjust until the base drum is dead center again (goniometer, correlation meter). Some simple conversion / effects can be used alike converting to mono and then back to stereo again, can keep the Basedrum dead center at all time during the mix. Specially working with bass drum samples, be aware they are straight in the middle centered. Beware of stereo, maybe even make the channel track mono (then you assure the signal is in the middle from the original signal). Specially the low range 50 - 120 Hz must be straight from center, sudden left or right events in this range are better to be avoided. Watch the correlation meter or goniometer with bass drums.

Frequency Range:

Find low kick bass drum for house in the range 60 Hz - 150 Hz.
 Find low kick for rock or pop up to 200 Hz - 300 Hz.

Cut, 0 Hz to 30 Hz (50 Hz), Reduction, Separation.
 Bottom, 60 Hz to 120 Hz, Find The Boom, kick or thumb.
 Around, 80 - 90 Hz, Solid Bottom (club) End.
 Cut, 80 Hz, 60's Records!
 Cut, 120 Hz to 250Hz. Muddy Lose it, Separation.
 Cut, 400 KHz, Open, Less Woody.
 Around, 1 KHz, Knock.
 Around, 2.5 KHz, Slap Attack.
 Boost, 2.5 KHz to 4 KHz, Kick Drum Definition Presence, Skin.
 Boost, 6 KHz, Click High End.
 Roll off, 10 KHz to 22 KHz, Trebles to set distance and reduction.

EQ: The Kick Thumb (head or hole) and the Skin are two basic frequency ranges to find. It can be very handy to create out of a single bass drum (sample or real) two signal tracks, one track with the 0-120 Hz frequency range and one track with the rest of the highs. This will make our purposes and plans with the bass drum more easy to adjust until they sound correct. The bass drum is most important to keep track of rhythm. The Skin will result in higher frequencies 2.5 to 5 KHz. Apply some boost or cut over here. Mostly a boost will make the Basedrum more rhythmic into the whole song or tracks, what is a good thing. Sometimes the Skin sound will progress towards 7.5 KHz. The Thumb or Head/Hole has bottom range between 60 and 100 Hz. Use a bell filter with a medium Q, hunt down the Skin and Thumb frequency ranges until you find the hotspots of both. Now you know what to reduce, and manage the two hotspot frequency ranges. The lower the bass drum the harder to edit, between 75 and 100 Hz is best main frequency. In clubs pressure develops at 90Hz, because of the speakers output. Below 60 Hz or deeper you have to be very careful and avoid swaying in panning, keep it centered at all time. Usually there is not enough high frequency content in a kick, so you should add it. Mostly in the range of 5 KHz to 8 KHz, sometimes 3 KHz and higher.

General Quality: Apply a little cut at 120-300 Hz and some boost between 60 Hz and 100 Hz (when needed, Thumb or Kick). The main frequency ranges are from 60 Hz to 100 Hz for bottom boom and 2.5 KHz to 5 KHz for the heads or the thump sound. Search in these area's to improve the quality of the base drum (boost when necessary). When the sound has a tendency to boom or resonate, try cutting between 200 Hz to 400Hz. Creating a modern sound, boost slightly in the 6 KHz to 12 KHz range, to accentuate the transient click when the beater hits the skin. All this to make the bass drum more clear and can be spotted by the listener for rhythmic understanding.

General Reduction : Apply a steep cut from 0 Hz to 30 Hz (50 Hz), adjust by listening to keep bass range or bottom end clear, but does not affect the boom thumb or kick (around 80 Hz). The Basedrum has a specific middle frequency between 60 to 100 Hz, for instance 80 Hz. Then you can be sure you can cut the Basedrum lower frequencies from 0 Hz to 60 Hz, this will leave some more room for the Bass to play. A lower midrange EQ cut, can help in the 120 Hz to 350 Hz misery area, Basedrum has no purpose here, so cut. For bass drum and bass thin out some 180 -250 Hz by a few db. Apply a mid-cut from 1 KHz to 3 KHz, where the Basedrum really does not need much power. Roll off some highs 10 KHz to 22 KHz that are not needed, this will affect distance also, set according to stage planning. Kick bass drum *house' 60 - 150 Hz, thumb humb. Kick bass drum usually rock or pop up to 200 - 300 Hz. Usually there is not enough high frequency content in a kick, so you should add it. Mostly in the range of 5 KHz to 8 KHz, sometimes 3 KHz and higher.

Compression: The compressor has two functions. First restricting dynamics (high level peaks, top end limiting, and compression) for the occasional over's. Second getting more punch trough transients, using a long compression attack time is more percussive and avoiding the sustain, meanwhile as a rhythmic aspect keeping bass drum in center localized. Use Opto mode for all percussive drums.

Side chain Compression: We could use side chain compression for more unmasking options on all instruments and effects The side chain compressor has found its way and application especially in house music and other styles as well. The Basedrum can be used for input to compress the Bass when the bass drum is playing its transient kick.

Gates : Sometimes a noise gate is used for this purpose in sync with the rhythm. A bass drum can be gated short then added an ambience reverb (from the drum group). A short gated base drum can sound good with a cut sustain and transients intact (short impulses are less tonal, so especially for Basedrum good to keep short). Also when the Thumb Kick frequency range is short, it will affect less loudness of the whole mix (lower frequencies have more power). The shorter you can make de bass drum the better, with the ambient reverb or room reverb the tail will be recreated and will be more clear rhythmically. Sync to tempo 32nd note when needed. If you need very deep notes or a very deep bass drum, remember the rule, the deeper the shorter.

Reverberation: Be careful and apply the rule ' less is more'. Too much will affect the Skin sound and therefore making less rhythmic inside the mix (masking, fluttering). Use no pre-delay. If needed stay below < 10 ms. A good ambient room or small room reverb / drum booth can be used for the whole drum group, so a good reverb is available already. Just send in some of the bass drum. When you have used sustain compression or gating, a reverb can help for setting some space and depth. The bass drum is usually left dryer treated with a short reverb, to stop it sounding indistinct and cloudy (muddy). Be careful with the reverb loudness, this will make the Basedrum flabby, rather than punchy and dynamic. Set reverberation pre-delay at zero to be in sync with the rhythm. An Ambience reverb is just enough (small reverb). A large reverb is almost never used; this will easily make the Basedrum reverb flabby, muddy and overcrowding. When a Basedrum reverb is switched off (bypassed) you must recognize the dry signal. When turned on, the reverb must add but not too much, but is enough to convey the 3d spatial information. Keep Basedrum reverb lowest in level according to all other reverbs used on other drum set instruments or tracks, even bass. Use no pre-delay or < 10ms to set the distance (drums are a bit behind the bass and main vocals according to our stage planning). You can maybe roll off some high trebles by EQ ing the reverb signal (> 10 KHz) instead. Remember we already rolled of some highs for reduction. Be sure the reverbed signal does not contain too much lows < 60 Hz to not affect the bass. Remember a small frequency ranged and shortened bass drum Thumb is best rhythmically. Basedrum needs the least amount of ambience reverb.


Panorama: The Snare belongs in the center (fundamental), according to the snare position (stage planning), maybe a little left or right (not much, very slightly). Beware of stereo, maybe even make the channel track mono, plugins can do this job and keeps the snare straight in center.

Frequency Range:

Cut, 0 Hz to 120 Hz, Reduction, Separation.
Between, 120 Hz to 400 Hz, Fatness Power, Wood.
Cut, 400 Hz, Snap.
Between, 400 Hz to 800 Hz, Body Thunk Sound.
Between, 800 Hz to 1.2 KHz, Power.
Cut, 1 KHz, Mellow.
Boost, 2 KHz, Bite.
Around, 4 KHz to 7 KHz, Crispness, Boxy.
Boost, 8 KHz, Sizzle.
Roll Off, 10 KHz to 22 KHz, Distance, Reduction.

EQ: Alike bass drum has two core frequencies, Lows 120 - 300 Hz, Strainer (high bands). Use a low cut at 80 Hz. Maybe pitch or tune the snare. Also splitting up the signal two ways makes processing easier.

General Quality: Anywhere from 120 Hz to 1 KHz, Boost or Cut. Boost around 240 Hz for more fatness, wood or power. Get some bite at 2 KHz. Crispness at 5 KHz, for more sizzle boost 8 KHz. Loose some boxiness at 6 KHz. For quality correction the ranges 110 Hz to 250 Hz (Bottom Snare) and from 3 KHz to 7 KHz are good boosting ranges. Accentuate the stick impact and rim shots at about 5 KHz. The rattle lies mostly between 5 KHz and 10 KHz. The bang is in the range of 1 KHz to 3 KHz. Body resonance can be found at 100 Hz to 250 Hz.

General Reduction: To separate Snare from the Basedrum (Frequency range collisions in dimension 2), use a steep EQ low cut up to < 120 Hz. A damping in the mid-range 1 KHz is where nice EQ cutting can be done to leave some headroom. Try applying some midrange cut to the rhythm section to make vocals and other instruments more clearly heard. Roll of some highs from 10 KHz to 22 KHz to set the distance according to stage planning. On digital systems often two components (samples) of a snare, actually produce the snare. The spectrum of the snare is the largest, cut steep at < 120 Hz. Snares resonate at 200 Hz - 300 Hz, cut and remove, it will be easier to place in a mix.

Composition and Tuning: A snare tuned to the chords or composition can be crucial. A snare with tonal content (tuned) can be more realistic and in tune with the song. You can be certain that a snare that is off tune (1 note plus or minus) can already sound horrible. Use the pitch to adjust the tonality (set the right toned snare). Some use pitch on all snare hits and adjust word by word throughout the composition. Again this may sound tedious and time consuming, still a pitch tuned snare is best.

Compression: First is top end limiting for the dynamic range (keep the transients intact but leave some headroom free). Second using compression with a long attack time for more transients (or maybe a gate). Creating more snappiness and percussiveness with longer attack times. Third, adjustment and control of the strainer sound (strings of the snare) with a faster release time. The snare will sound best when the transients are loud and of good quality. Using a gate or compressor to wipe away the sustaining snare sound is perfectly correct (especially in sync with the tempo). Use Opto mode for all percussive drums.

Gates: All right we tend to give the snare a good wide big reverb to make an open sound. Short snares are very nice, especially when going into the big reverb, so we need a gate to just only allow the transients and important sounds of the snare through. We can also cut the snare sample by manual edits. Mostly then mixed into an ambience reverb (group) for an ambience result. Maybe place a noise gate after the reverb device. For longer snares sync the gate to tempo.

Reverberation: Without a sustaining snare, we can now choose a decent reverb and make snare sound the way we need. For the snare to be different, single track a medium sized Room Reverb. It is perfectly all right when you choose a large reverb for the snare alone (a way bigger, larger room, then for instance we use on the bass drum or rest of the drum set). To separate the snare from the other drums a large reverb will help it stand out more or a short crisp plate program. Snare drums are traditionally treated using a plate reverb, hall settings also work well. Try 0.5 sec for a short reverb, and over 2 secs for an obvious effect. Use no pre-delay or < 10 ms, checking the rhythmic (use a high treble’s roll off for setting distance instead). Place a gate after the reverb and bring it within the beat of the snare (sync). If you don't roll off the highs of the snare or this reverb, the snare will sit nicely upfront (just cutout the lows). Place the snare a little behind the Basedrum, at its natural placement.

HI hat.

Panorama: The HI hat can be placed slightly left or right, according to its natural position more right. As the rest of the drum kit is all not fundamental, we place them into the panorama according to our stage plan. The HI hats needs to have a low-cut filter < 250 Hz. The more deep frequencies in the hi hats origin, the more you place it right (outwards). HI hats slightly to the right, place the shaker left (Counter weight instruments).

Frequency Range:

Cut, 0 Hz to 200 Hz (500 Hz), Reduction, Separation.
Boost, 800 Hz, Fullness.
Cut, 1.5 KHz, Smoothness.
Boost, 4 KHz to 5 KHz, Edge, and Crispiness.
Boost, 10 KHz, Sparkle.
Cut, > 15 KHz, Roll Off, Distance, Reduction.

Quality: Looking for some Fullness, Smoothness, Edge and Sparkle, cut or boost. This all depends on the actual sound of the HI hat, most usually they dominate the 8 KHz to 12 KHz area, so first apply a boost 3 dB and then surf the area until you find a sound which is suitable for the mix. Main frequencies are the ring from 7 KHz to 10 KHz. The stick noise at about 5 KHz, and a clang in the range of 500 Hz to 1 KHz. Use an oversampling EQ of quality.

Reduction: Apply a steep filter cut from 0 Hz to 200 Hz (500 Hz). Depending on the kind of HI hat, cut a lot more to about 3 KHz (5 KHz). However hi hats can use frequencies as low as 5 KHz, these don't necessarily contribute to the sound, they only serve to take up space in a mix (headroom). Roll off for distance > 15 KHz according to stage planning.

Compression: Do not use a gate on hi hats. Mostly HI hat signals are not compressed at all, or only slightly when some dynamical events need to be reduced or gained, use note or event based manual edits or automation to correct those parts. The HI hat has no natural dynamic content in the lower frequency ranges, so has not much effect on the dynamics, though is ear able.

Reverberation: Use no pre-delay or < 10 ms, checking the rhythmic (use a high treble roll off for setting distance instead). HI hats work well with a short to medium bright reverb setting. Try adding a high level of early reflections (ambience) to add interest and detail. Dance music uses little or no HI hat reverb to retain the timing and impact of the dry sound (else sync to tempo). Try a short crisp plate program.


Panorama: Keep centered, maybe use a stereo expander to set as wide as possible. Or used as an counterweight for other instruments (especially the drum set).

Frequency Range:

Cut, 0 Hz to 200 Hz (400 Hz), Reduction, Separation.
Cut, 1 KHz, Openness.
Boost, 12 KHz, Zing, Air.

Quality: Add some lusture around 4 KHz with a high shelf EQ. When processing highs use a quality or oversampling EQ.

Reduction: Apply a low-cut filter from 0 Hz to 200 Hz (400Hz). Roll off some high trebles to send the overhead to the back of the stage (according to your stage plan).

Reverberation: Use no pre-delay or < 10 ms, checking the rhythmic (use a high treble’s roll off for setting distance instead). Try a longer delay time > 1.2 S, using a hall programm with a decay of about 1.5 seconds.

Make Your Drum Overheads Sound Amazig, compressing the drum overheads is a great way to make your drums pop. You can tame any unwanted transients with the attack and release times. U can really smooth out and make the drums more consistent and make your drums sound a lot better. If your going for a heavier drum sound, you can really brick wall compress the drum overheads and get a really juicy sounding drum sound. Really harsh ratio and threshold setting can make the cymbals ring out for ages combined with a long release time. Sidechaining the overheads to the kick drum can really make the drums pump and breath, giving your mix a lot of life and energy.


Panorama: Place according to stage position normal cymbals (not crash cymbals) need to be close to the HI hat on the right side, either more left or right. Sometimes at center but widened alike the overheads or used as counterweight. Crash cymbals can be placed anywhere, be sure to pan as wide as possible.

Frequency Range:

Cut, 0 Hz to 100 Hz (400 Hz), Reduction, Separation.
Boost, 100 Hz to 300 Hz, Clunk stick, Clang or Gong.
Cut, 200 Hz to 400Hz, to thin cymbals, Separation.
Cut, 1 KHz, Openness, Boxy.
Cut, 1.5 KHz to 6 KHz, Ring.
Boost, 7.5 KHz to 12 KHz, Shimmer, and Sizzle.
Boost, 12 KHz, Zing.
10 KHz to 16 KHz, Air, Crispy Cymbals.
Roll Off, > 12 KHz, Limiting, Distance.

Quality: Main clang or gong sound at 200 Hz, crispness at 5 KHz. There are many types of cymbals, splash, china, effect cymbals, orchestral cymbals, marching band, gongs and specialty stuff. Therefore only adjust the cymbals marginally. Reducing is always better when you adjust cymbals. Cymbals can be overcrowding and irritating. The main resonance lies below 1 KHz, in the range from 75 Hz to 300 Hz. From 1 KHz to 3 KHz is the bang of the beat and from 5 KHz to 10 KHz is the click. Resonance is from 8 KHz to 15 KHz. Pay attention to the quality of plugins used, adding some brilliance. When processing highs use a quality or oversampling EQ.

Reduction: Cut anywhere from 100 Hz to 350 Hz. A gentle roll off in lows 12 to 24db so it does not phase with the snare.

Compression: Do not try using a gate. Mostly signals are not compressed at all, or only slightly when some dynamical events need to be reduced or gained, use note or event based manual edits or automation to correct those parts. Has no natural dynamic content in the lower frequency ranges, so has not much effect on the dynamics, though is ear able. For crash cymbals their frequency of play is often very small, therefore compression is unsuitable and can be a hassle when trying to setup. Alike toms when they are just sporadic, we tend to only use compression when the input signal is more constant over the timeline included and can be managed more. Else use manual edits instead.

Make Your Drum Overheads Sound Amazing, compressing the drum overheads is a great way to make your drums pop. You can tame any unwanted transients with the attack and release times. U can really smooth out and make the drums more consistent and make your drums sound a lot better. If your going for a heavier drum sound, you can really brick wall compress the drum overheads and get a really juicy sounding drum sound. Really harsh ratio and threshold setting can make the cymbals ring out for ages combined with a long release time. Sidechaining the overheads to the kick drum can really make the drums pump and breath, giving your mix a lot of life and energy.

Reverberation: The cymbal track is already a kind of close ambient room sound. Use no pre-delay or < 10 ms, checking the rhythmic (use a high treble’s roll off for setting distance instead). Cymbals particularly can be enhanced by a longer reverb.


Panorama: Hi Tom placed slightly or far Right, Low Tom placed far left, or opposite. Remaining Toms placed in between, alike their natural stage. Mid tom slightly left, floor tom far left.

Frequency Range:

Cut, 0 Hz to 30 Hz (50 Hz), Less Muddy Mix events, Separation.
Between, 80 Hz to 300 Hz, Fullness, Boom.
Boost, 400 Hz to 800 Hz, Warmth.
Between, 1 KHz to 3 KHz, Ring.
Between, 3 KHz to 8 KHz, Attack.
Between 8 KHz to 16 KHz, Air, Distance.

Quality: If the toms sound weak, use a bell EQ about 100 Hz to 200 Hz (150 Hz mid frequency) or identify the exact center frequency for each tom. Rack toms fullness at 240Hz, attack at 5 KHz. Floor toms fullness at 80-120 Hz, attack at 5 KHz. Each tom has only one frequency range, so this can be adjusted with only one single steep bell filter, toms mostly have just a small frequency range sweet spot. Roll of some trebles to set the distance.

Reduction: Cutting out the lower bottom end < 120 Hz or more can free up some headroom, but as toms are not continues events maybe < 50 Hz is quite ok, to keep some power.

Compression: A noise gate on the toms can remove some sustain or some compression artifacts, keeping the transients intact. Sometimes a manual edit, does not take any more time as toms only appear in sudden events. Use Opto mode for all percussive drums. Alike crash cymbals when they are just sporadic, we tend to only use compression when the input signal is more constant over the timeline included and can be managed more. Else use manual edits instead.

Making The Toms Punch, compression on toms can create some amazing results. Using heavy enough compression along with a gate can make your tom drums seriously punchy. Even if you don't have individual tom mics and just an overhead pair, or just a single overhead mic, compression can really make the toms punch out. Think of songs like Shine On You Crazy Diamond by Pink Floyd. The compression on the toms make them really punchy and beefy, really adding to the mix.

Reverberation: For toms maybe a large snare reverb (same as snare) can bring them out. Toms have a natural sustain, so don't need much reverb. Plate and small room settings are good for pop, with metal benefiting from longer settings. Hall settings are good for a big tom sound or try a short crisp plate program.


Panorama: There is no basic placement for panning on percussion and cymbals. Percussive elements are often panned left or right and kept away from the center, set as much outwards as possible. A stereo expander can bring the percussion elements even more outwards. We like to pan percussion more left or right, not centered. Bongo’s to the left and far behind in distance, conga's far to the right and also far behind in distance. Panning outwards they remain unmasked by other signals, set in distance when other instruments already overcrowd the stage.

Frequency Range:

Cut, 0 Hz to 120 Hz, Reduction, Separation.
Cut, 200 Hz to 400 Hz, Higher frequency percussion.
Between, 200 Hz to 240 Hz, Resonance.
Around, 5 KHz, Presence Slap.
Between 10 KHz to 16 KHz, Air, Crisp Percussion.

Quality: Resonance at 200 Hz to 240Hz, presence slap at 5 KHz. For distance depth roll of some high trebles, send the more to the backstage.

Reduction: Roll off some lows from 0 Hz to 120 Hz or more, as they are outwards and do not need a lower transmission (panning laws). According to stage placement, roll off lower frequencies. Percussion and cymbals can be cut 1 KHz and higher. Cut with shelf filter at 800 Hz and higher (1 KHz - 4 KHz), be careful otherwise will sound acid and unnatural.

Compression: Compression can help bring forward the transients while reducing the sustaining sounds (keep some headroom). Use Opto mode for all percussive drums.

Chorus: Often used.

Reverberation: As for using reverbs or delay, maybe the reverb placed for the snare (Group, Send)can be useful also for percussion purposes, we tend not to use the ambient reverb of the whole drum set, maybe just some to glue into some togetherness with the rest of the drum set. Percussion requires a long reverb with little pre-delay and a little high frequency cut or is damped by the reverb setting. For the Percussion (Group, Send) a medium sized Room from 1.5 seconds to 2 seconds of delay time. A pre-delay of 15 ms and a medium roll off in frequency (damped or EQ). The masking effect might hide the reverb, but just set the loudness of the reverb high enough to get some 3d spatial information transmitted. Maybe a stereo expander after the reverb signal and some automation will solve the hiding problem, panned outwards, just watch the correlation meter as you are using the stereo expander to widen. A delay can help sweeten the percussion, only when you adjust the delay in tempo synced and keeping pre-delay short. Reverberation, use no pre-delay or < 10 ms, checking the rhythmic (use a high trebles roll off for setting distance instead). Transients are more important for percussive sounds, they are mostly placed backstage so for rhythmical content we need the original transients to be heard. Percussion instruments are placed consciously toward the rear, we will need a large reverb with some pre-delay and filtered trebles.  Reverb can be generously applied here, so the masking effect will stay away or is overpowered by reverb and brings over the 3d spatial information. Reverb layering for instance, gives percussion tracks a medium, thick room with quality. The return is processed with a little widening to counteract the masking effect and to place the percussion behind the drums. A little pre-delay on the reverb and slightly attenuated or lowered trebles.


Level: Bass must sound in the mix. While playing single mode you cannot hear the outcome inside the mix. Mostly bass is 2 or 3 dB higher on the vu-meter than the bass drum. Use Envelope, Filter Release and Sustain, Amp Release and Sustain for creating a bass on a synth, try to fiddle with envelope ADSR as well as above mentioned parameters. The levels between bass drum and bass should be, bass drum is -1 dB or -3 db on the vu-meter. Usually the bass drum will more or less disappear. We can cut some frequency range from the bass, just sweep around 60 - 150 Hz and at one moment the kick will come back (thumb sub kick will return and be ear able again). Also choose the kick and bass its timbres. As another resolvement we could delete all notes from the bass that overlap the kick, mainly 1/2/3/4 measured kick overlapping (start of the rhythmical bass drum content, midi, note deleting or sample cutting). Pay attention to the duration of a kick, it should correspond to the speed of the track. A beginners mistake will be just raising the kick at 60 - 120 Hz, this will raise the bass drum but only will be of more level, not consistency.

Panorama: The Bass is most fundamental (next to the Basedrum). A bass that is not played in center entirely through the track timeline, but rolls a bit from left to right is offsetting the balance of the mix, because the bass uses heavy lower frequency components. Bass is always placed at the center and if not, it is correction time. So really keep the bass dead centered. Sudden events in the bass that are left or right, make the bass less effective, sway around making transmission of both speakers less effective. Maybe convert bass to mono or use a mono convert at the end of the channel. Convert bass samples to mono. The bass needs to be dead center at all times! Use a correlation meter and most of all the goniometer to make the bass dead center in all events.

Frequency Range:

Bass Note Range: 33 Hz (C1) to 523 Hz (C5).
Bass Guitar Note Range: 31 Hz (B-1) to 392 Hz (G2).

Roll Off, 0 Hz to 30 Hz, Reduction, Separation.
Boost, 60 Hz to 100 Hz, Bottom, Small Frequency Range, Careful and Exact.
Cut, 60 Hz, Humming Noises, Eliminate.
Boost, 100 Hz to 120 Hz, Pointy, Prominent, Fat.
Between, 120 Hz to 300 Hz, Warmer.
Around, 200 Hz, Leave it or cut.
Around, 250 Hz, Nasty Bass Frequencies, Cut.
Between, 400 Hz to 800 Hz, Clarity.
Between, 500 Hz to 1.5 KHz, Pluck noise.
Around, 800 Hz, Mid Tops, Fret noise.
Around, 2 KHz, Presence and Definition.
Between, 2 KHz to 6 KHz, Edge, String noise.
Roll Off, > 10 KHz, All Highs, Distance, Reduction.

Reggae Bass Sound : Boost, 40 Hz, 10 dB, Boost, 80 Hz, 12 dB, Cut, 160 Hz, - 8 dB, Cut, 240 Hz, - 6 dB, Cut, 600 Hz, -15 dB, Boost, 1 KHz to 1.5 KHz, 1-3 dB.

EQ, Frequency wise the bass needs room. Every low frequency 0 - 120 Hz content must be bass only, especially in center. Only the small range 80 - 100 Hz Basedrum bottom kick frequencies are welcome, other instruments should have a big cut over here.

Quality : To get some quality maybe a bit of a boost in the 40 Hz to 70 Hz range, better not to do so, be careful. For that wooden sound try the 750 Hz to 1 KHz range. Main area's bottom at 60 Hz to 80 Hz, attack pluck at 750 Hz to 1 KHz. 800-1200 Hz is nasal, or the woody part. String noise pop at 2.5 KHz.

Reduction: Bass needs space to play, any unnecessary sound event in the lower frequency range will create a muddy bass (masking). So it is best to cutoff rest of instruments frequencies anywhere from 0 Hz to 120 Hz at least, but we pay special attention to the rest of fundamentals, Bass drum, Snare and Main Vocals. For bass this does mean a lot, we expect that the range 0 Hz to 30 Hz can be cut steeply, while leaving 30 Hz to 120 Hz (180 Hz) intact. The only instrument that can go as low as 30 Hz is the Bass, no other instruments will get this low. So for Bass you can cut from 0 Hz to 30 Hz. Just to get rid of all flabbiness, pops, rumble, bottom end bass sounds. A cut or damp in the 120 Hz to 350 Hz (500 Hz) misery range, will get some more headroom back, listen how much you can cut over here. For bass drum and bass thin out some 180 -250 Hz by a few db. Roll off some highs > 10 KHz, to set the distance according to stage planning. Also the bass does not really need any high frequencies, so cut anyway. The bass must fall behind the main vocals in distance. Try to cut 150 Hz - 400 Hz on bass, often this bass part is resonant and unwanted, it keeps the misery area more clear. For better earabillity a bass can be raised a bit at 1 KHz to 1.5 KHz.

Compression: Control balance of heavy notes and stop notes, damping or dead notes. Dynamic limiting of sudden level peaks and irregular playing. Creation of sustain with long notes (especially in song with low tempo), with a long release time, sync to tempo. Boosting quieter side notes (funk with fast release times must exclude sustain). Supporting rhythmic and percussiveness with long attack times (transients). Sometimes multiband compression can help a bass, only resort when all else fails. A good played baseline is worth millions, making easy manual level / volume / muting adjustments and less compression tricks needed for avoiding those dead notes or just edit them out of the baseline. Bit slower attack and slower release, so you leave or accentuate the transient of the hit. Off course if you want to smooth out the bass and bury your bass in the track by getting rid of the attack, have a fast attack on the compressor. Attack time can be set to between 10 ms to 40ms. For getting Bass more sustain, ratio 4:1 (to 6:1), reduce threshold until it hits. View the gain reduction meter, try to get the bass as stable in sustain as can be. Use some gain to compensate for reduction. Set release, so that the sustain reduction is stable. Attack times that are longer 5 ms to 30 ms is letting the transients (first part of a note) through pass. The compressor attack time can be used to control the snappiness and therefore the definition at the start of each note. The amount of compression controls the balance between the heavy sounding notes and corrects damping sounds and dead notes. The dead notes can be well emphasized with the compressor by sustaining. A short attack time cuts the transients more, therefore raising the sustain. The length of the bass note is the groove. Baselines specially sound good when notes sustain equally on each second or fourth quartertone.  A compressor can also be used to shape dynamics, limiting peaks. (Creating sustain for long sustain tones in tempo to the rhythm). A long release time (according to tempo) is more rhythmical.  If the bass sounds weak in the transients you can support a long attack time. The release time must be set for the bassist playing style, short notes will need a fast release time. Watch for background noise pumping.

Side chain Compression: We could use side chain compression for more unmasking options on all instruments and effects The side chain compressor has found its way and application especially in house music and other styles as well. The Basedrum can be used for input to compress the Bass when the bass drum is playing its transient kick.

Reverberation: Try not to use. Else try an ambient reverb (alike the base drum on the drum group) and use very subtle or inaudible. Maybe a small Room or Ambient reverb. With a bit of pre-delay from 0 ms to 10 ms (0 ms please for rhythmical content), when set higher synced to tempo. Just roll off some trebles after the reverb to set the stage again. Use no pre-delay on reverbs, (maybe a bit just to set the distance behind the main vocals, but you could use a treble roll off instead). Bass just needs enough ambient reverb, just slightly more than the bass drum has.

Chorus: Double signal the bass, splitting into two signals, lows and highs. On the highs the chorus can do its job without phasing > 250 Hz (no to sway the lower frequencies around, keeping it centered). Chorus and phase, on a bass only above > 250 Hz. Maybe you have split the bass signal already in two frequency parts, then just use chorus on the higher part > 120 - 250 Hz.

Guitar Acoustic.

Panorama: In mixes guitars mostly come in pairs. As we can use this technique to set one left and one right, to keep a balanced feel and to stay upfront. As guitars might be understood as crucial, they are not fundamental. So not placed in center. When only a single guitar is played, maybe use some other instrument on the opposite side (counteract). When acoustic guitar and vocals are the only fundamental mix components, we have to use different mixing techniques, vocals centered upfront and acoustic guitar off-center with a widening effect more outwards. Even for solos a switch filter can become a good tool. Keeping the main vocal upfront at center, we try to avoid masking the main vocals.

Frequency Range:

Cut, 0 Hz to 80 Hz (120 Hz), Reduction, Separation.
Between 80 Hz to 120 Hz, Power.
Between 200 Hz to 400 Hz, Boom, Warmth.
Cut 500 Hz to 1 KHz, Brittleness.
Between 1 KHz to 1.5 KHz, Strumming.
Between 2 KHz to 3 KHz, Abrasion, Bite.
Between 2 KHz to 5 KHz, Clarity, Honky.
Between, 5 KHz to 10 KHz, Nasal.

Quality: Checkout the highs with a spectral analyzer. Add Sparkle, try some gentle boost at 10 KHz using a Band Pass Filter with a medium bandwidth. Bottom at 80 Hz to 120 Hz, body at 240 Hz, clarity at 2.5 KHz to 5 KHz. Apply a little cut at 300 Hz. Roll off a bit of the high trebles to set the distance. Guitars solo-ed can sound thin, while in a mix they are good.

Reduction: Cut lows from 0 Hz to 80/120 Hz (250 Hz/400 Hz) depending on the note range and when the guitar events are combined with vocals (automation). Be sure everything below 100 Hz is at -15dB cut. A good steep roll off in EQ from 0 Hz to 120 Hz (250 Hz) can help free up some headroom and get rid of some nasty guitar body sounds. Now the correlation gets better. Also when the highs are not nicely rolled of, do so with a roll off in EQ for distance. Sometimes the lead vocals and guitar play at the same time, meaning they need to stay in distance upfront. We can try to switch cutoff filters when the main vocals are sounding ( > 250 > 400 Hz). Use a quality oversampling EQ on the range > 8 KHz. Cut low guitar frequencies with a shelf up to 400 Hz, a guitar can have lows from 84 Hz, a bass can have lows from 30 Hz, so they tend to overlap. You can add frequency to the guitar from 1.2 KHz to 3 KHz, when it sounds to acid cut.

Compression: Can sound weak when adjusted with an EQ in front of the compressor. The frequency range and broadband was specified by the EQ and removes low rumble and other unwanted signals. Attack time can be set to between 10 ms to 40ms, for more transients and to be percussive/rhythmic. Fast attack usually somewhat fast release. Compress with ratio 4:1, with an attack of about 5 ms, hold 250 ms, release 50 ms (100 ms to 250 ms), for the sustain sounds. Make up for gain. You might even need to add an effect that generates more warmth. Uncompressed guitars are difficult to handle inside the mix. For a more percussive rhythmic approach use Opto mode, for a softer more contained sound use RMS mode.

Reverberation: A short bright plate reverb can work well on a steel strung acoustic guitars. Apply some reverb or delay (or any other guitar effect) can help to counteract as opposite. Many guitar players prefer the sound of the spring reverb in their amplifiers. Maybe setup a delay afterwards. Maybe use some small doze ambient reverb available on a group or available send. Use a hall reverb for starters.

Delay: Delay can work out better for guitars that must stay upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make the main guitar upfront but still have some space. Maybe an widener or expander placed behind.

Guitar Electric.

Panorama: In mixes electric guitars mostly come in pairs. As we can use this technique to set one left and one right, to keep a balanced feel. As guitars might be understood as crucial, they are not fundamental. So not placed in center where it is already overcrowded. Don’t be shy with panning settings, more outwards. When only a single electric guitar is played, maybe use some other instrument on the opposite side (counteract). When acoustic guitar and vocals are the only fundamental mix components, we have to use different mixing techniques, vocals centered upfront and acoustic guitar off-center with a widening effect more outwards. We can use a switch filter setting when main vocals are sounding or not. Even for solos a switch filter can become a good tool. Keeping the main vocal upfront at center, we try to avoid masking the main vocals.

Frequency Range:

Cut, 0 Hz to 80 Hz (120 Hz), Reduction, Separation.
Between, 125 Hz to 250 Hz (400 Hz), Warmth.
Boost, 500 Hz Body.
Cut, 500 Hz to 1 KHz Brittleness.
Boost, 2 KHz to 3 KHz Abrasion, Bite.
Filter, 2.5 KHz 3db LF / 6db MF.
Crisp, 3 KHz to 5 KHz.
Roll Off, 4 KHz to 4.5 KHz, Irritating.
Boost, 6 KHz, Distorted Guitars.

Quality : Fullness at 240 Hz, bite at 2.5 KHz. Clean electric guitars can be treated like acoustic guitars. Apply a little cut at 300 Hz. Roll off a bit of the high trebles to set the distance.

Reduction: Cut lows from 0 Hz to 120 Hz (250 Hz) depending on the note range and reduction needs (are the main vocals sounding?). Be sure everything below 100 Hz is at -15dB cut. Now the correlation gets better. Also when the highs are not nicely rolled off, do so with a roll off in EQ for distance. Sometimes the lead vocals play guitar at the same time, meaning they need to stay in distance upfront. Checkout the highs with a spectral analyzer. Add Sparkle; try some gentle boost at 10 KHz using a Band Pass Filter with a medium bandwidth, use oversampling quality EQ.

Compression : With ratio 4:1 with an attack of about 7 ms, hold 250 ms, and release 50 ms. Fast attack usually somewhat fast release. When needed enhance the transients, electric guitars tend to have enough sustain. Make up for gain. You might need even need to add an effect that generates more warmth. For sustain, set a fast attack time and a release of around 250 ms, when really needed. Set the ratio from 4:1 upwards, and apply gain reduction up to 12 dB.

Controlling Guitar Dynamics, When recording lead guitar there are always a few notes here and there really jumping out a lot louder than the rest of the guitar track. Usually compress with a ratio of about 5:1, then turn the threshold down until you can hear the audio being squeezed a bit. Then set the attack time so the transients are shining through unaffected and the rest of the signal is getting compressed, ultimately making the audio more consistent dynamically. Try the release settings until it fits the song.

Reverberation: Apply some reverb or delay (or any other guitar effect) can help to counteract as opposite. Many guitar players prefer the sound of the spring reverb in their amplifiers. Maybe setup a single delay. Use a hall reverb for starters.

Delay: Delay can work out better and makes any instrument stay more upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make it stand upfront but still have some space (unmasking). Metal guitars are often panned completely left or right and sometimes make use of heavy delay.

The huge modern rock and metal guitar sound - There is no one answer, but there are some very typical things used to create great modern rock guitar sounds. Often, it's a tasteful guitar going into an all-tube 1x12, with different microphone positions and types being tried out and the best one selected (many rock engineers prefer ribbon microphones). The microphones are amplified by a great preamp, and may go through a "color" compressor (often a high quality tube compressor). Finally it is converted with a high quality AD/DA converter as they the sound is recorded into the DAW. There may be several overdub layers recorded and mixed in subtly. Many of the other tips suggested here are in use.


Panorama: Also it is more likely we will place the piano (as an not fundamental instrument) by setting pan more left or right. When it appears as fundamental instrument (alongside main vocals) we can try to widen and expand around the main vocals. We could counteract the piano with another instrument or reverberation device opposite of the panorama. Or even use a widened reverb tail that progresses outwards. A piano can get any placement, when played by band members, left or right. But however sometime the main vocals will also play the piano, so therefore maybe set the main vocals a bit to the left or right, with the piano as opposite. In this case we do not roll off any trebles and keep them upfront. We can also place the piano slightly behind the main vocals at center (as fundamentals), switch mute the filter of the piano in solo mode or when main vocals are sounding.

Frequency Range:

Piano Note Range: 28 Hz (A-1) to 3651Hz (B7).

Cut anywhere from 30 - 120 Hz, Reduction, Separation.
Boost, 100 Hz, Power.
Around, 250 Hz, Clarity.
Boost, between 1 KHz -3 KHz, More Aggressive.
Boost, 2 KHz, Harmonics.
Boost, 6 KHz, Attack.
Boost, 12 KHz, Sparkle, Air.

Quality: Bottom at 80 Hz to 120 Hz, presence at 2.5 to 5 KHz, crisp attack at 10 KHz, honky-tonk sound (sharp Q) at 2.5 KHz. Roll off some highs when needed for distance.

Reduction: Difficult to master inside a mix. Basically because piano can play a wide range of notes on all octaves. Depending mixing purposes, we can address two situations. First we have a mix where already Basedrum and Bass are fundamental instruments playing. In this case the piano becomes not fundamental. For the Bass range to stay clear, we can cut a lot from 0 Hz to 120 Hz out of the bottom end of the piano. Second, when we have no Basedrum or no Bass playing, or both are absent, thus the piano can be more fundamental, we can leave some lower frequencies inside the spectrum and be more careful rolling of the lows. Anyway a good EQ cut from 0 Hz to 30 Hz (50 Hz) is always applied. Still we like to roll off all frequencies lower then < 120 Hz. We can roll off some high trebles for distance.

Reverberation: Depending on stage planning, we can add some ambience reverb for upfront, and depth by using a larger reverb. Maybe use some delay instead (especially when needed upfront).

Delay: Delay can work out better and makes any instrument stay more upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make it stand upfront but still have some space.



Panorama: An Epiano can get any placement, when played by band members, still not at the already overcrowded center. It is more likely we will place the Epiano (as an not fundamental instrument) by setting pan more left or right. We could counteract the Epiano with another instrument (piano) or reverberation device opposite of the panorama. However when fundamental sometimes combined with the main vocals. So therefore maybe set the main vocals a bit to the left or right, with the Epiano as opposite. In this case we do not roll off any trebles and keep them upfront. We can also place the Epiano slightly behind the main vocals at center (as fundamentals), switch mute the filter of the piano in solo mode or when main vocals are sounding.

Quality: A famous Epiano is the Rhodes Epiano. Less difficult to master inside a mix. Roll off some highs when needed for distance.

Reduction: Depending mixing purposes, we can address two situations. First we have a mix where already Basedrum and Bass are fundamental instruments playing. For the Bass range to stay clear, we can cut a lot from 0 Hz to 120 Hz out of the bottom end of the Epiano. Second, we have no Basedrum or no Bass playing, or are both absent, thus the Epiano can be more fundamental, we can leave some lower frequencies inside the spectrum and be more careful rolling of the highs (keep them upfront). Anyway a good EQ cut from 0 Hz to 30 Hz (50 Hz) is always applied. Still we like to roll off all frequencies lower then < 120 Hz when not fundamental.

Reverberation: Depending on stage planning, we can add some ambience reverb for upfront, and depth by using a larger reverb. Maybe use some delay instead (especially when needed upfront).

Delay: Delay can work out better and makes any instrument stay more upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make it stand upfront but still have some space.


Panorama: Organ alike piano and Epiano can combine together, maybe placing organ left and (e)piano right. Counteracting is common, placing left or right. Remember that organ when using a rotary effect (Leslie effect) can move inside the panorama. Keep track of where the organ is supposed to be placed according to your stage planning.

Frequency Range:

Roll Off, 300 Hz, Lowers Power.
Boost, 2 KHz to 3 KHz.

Quality: Bottom at 80 Hz to 120Hz, body at 240Hz, presence at 2.5 KHz. Roll off some highs for distance.

Reduction: For the Bass range to stay clear, we can cut a lot from 0 Hz to 120 Hz out of the bottom end and bass range.

Reverberation: Depending on stage planning, we can add some ambience reverb for upfront, and depth by using a larger reverb. Maybe use some delay instead (especially when needed upfront).

Delay: Delay can work out better and makes any instrument stay more upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make it stand upfront but still have some space.


Panorama: More left or right, we use counteracting and our stage plan to decide where to place these instruments. Keyboard often sweep in panning, just keep them out of center, reduce the lows. More left or right but maybe use some stereo expander when they are more fundamental.

Quality: Keyboards usually can play lots of different instruments altogether. Keyboards always use a low-cut filter anywhere between 50 - 150 Hz. Lookout for DC offset and remove it.

Reduction: Dividing each keyboard instrument and giving them a separate track makes the mix more adjustable. Depending on the instrument played by the keyboard, decide what to do. Playing a Bass or Piano, Epiano or Synth, Brass, with a keyboard is deciding where their frequency ranges are. Cutting still from 0 Hz to 50 Hz is always applied, but still leaving the Bass range alone we should cut towards 120 Hz or even higher. Also we could control the trebles to set some distance according to individual stage planning of each instrument. Sometimes keyboard can play bass or Basedrum (drums), then we will refer to them as bass and drums and react alike (making then fundamental instruments). Sometimes keyboards can play percussive instruments; we react accordingly as if they were real percussive instruments.

Reverberation: For Background Keyboards or background sounds (Group) use a Large Reverb with a pre-delay of about 25ms (Check the snare reverb for starters). Use an EQ to roll off the highs strongly and the reverb sends all to the back in distance. Maybe a Modulation delay. Use a hall reverb for starters.

Delay: Delay can work out better for keyboard that must stay upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make the keyboards upfront but still have some space.


Panorama: Panned more left or right, we use counteracting and our stage plan to decide where to place these instruments. Synths can play bass sounds as well, so decide on tactics according the instrument sound. For a good wide sound on synths and guitars, use a double track recording, record twice and pan left and right, the slight nuances in playing make it very wide.

Quality: There is no best synth. Synthesizers usually can play lots of different instruments altogether, mostly analog or digital artificial sounds. Synths, leads (100 Hz cut), pads (400 Hz cut) and strings (1 KHz cut). Cut high frequencies of every synth instrument (12 KHz - 20 KHz).

Reduction: Dividing each instrument and giving them a separate track makes the mix more adjustable. Depending on the instrument played, decide what to do, deciding where their frequency ranges are. Cutting still from 0 Hz to 50 Hz is always applied, but still leaving the Bass range alone we should cut towards 120 Hz or even higher. Also we could control the trebles to set some distance according to individual stage planning of each instrument. Sometimes a synth can play bass or Basedrum (drums), then we will refer to them as bass and Basedrum and react alike (making then fundamental instruments). Sometimes a synth can play percussive instruments, we react accordingly as if they were real percussive instruments.

Compression: Most synthesizers don't need compression, be scarse. Analog filter sweeping can be compressed for peaks with a ratio of 4:5 to 6:1.

Reverberation: Adding delay can support the synth sound to become more natural and fitting inside a mix, used as a creative effect. To set the distance we could roll off some highs. Maybe a Modulation delay. To thicken synth sounds, try a bright reverb with predominant early reflections. A short, high level reverb makes the synth sound like multiple instruments in an acoustic space. Use a hall reverb for starters.

Delay: Delay can work out better and makes any instrument stay more upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make it stand upfront but still have some space.


Panorama: Depending on the frequency range of alt-violins and the higher violins we place them more outwards (according to panning laws. As for modern pop music we might set all strings behind the drummer, spreading them out in panorama (stereo expander). Leaving the viola’s and cello’s more centered. The violin and alt-violin more outwards. For a more classical approach use the orchestral stage plan to place all stringed instruments.

Frequency Range:

Violin Note Range: 195 Hz (G3) to 3136 Hz (G7).
Viola Note Range: 131 Hz (C3) to 2093 Hz (C7).
Cello Note Range: 65 Hz (C2) to 1047 Hz (C6).

Roll Off, 0 Hz to 120 Hz (180 Hz), Reduction, Separation.
Around, 200 Hz to 500 Hz, Fullness.
Cut, 800 Hz to 1 KHz, Recession.
Boost, 5 KHz to 10 KHz, Clarity.
Between, 7.5 KHz to 10 KHz, Scratchiness.
Between, 10 KHz to 16 KHz, Air, Sparkle (if present).
Roll Off, 10 KHz, Distance, Reduction.

Quality: Fullness at 240 Hz, scratchiness at 7.5 KHz to 10 KHz. Roll off highs for distance.

Reduction: Cutting a lot of lower frequency range 0 Hz to 120 Hz (195 Hz), but not harming the main frequency range. Viola's and cello’s have a lower frequency range, so we might cut a little less 0 Hz to 120 Hz. Still we like to cut all rumble and lower frequencies, also rolling off highs to set them behind the drummer.

Reverberation: High pre-delay for strings can send them assigned to the back, to the back rows and roll off the highs of the reverb can set even more backwards.


Panorama: Horns, Trumpets, Trombones and Tuba. Depending on their frequency range and placement, decide where they fit in. Scattered across the whole panorama. Placing lower instruments more centered and higher instruments more outwards (panning laws).

Frequency Range:

Trumpet Note Range: 115Hz (E3) to 1047 Hz (C6).
Trombone Note Range: 82 Hz (E2) to 698 Hz (F5).
French Horn Note Range: 65 Hz (C2) to 698 Hz (F5).
Tuba Note Range: 37 Hz (D1) to 349 Hz (F4).
Piccolo Note Range: 587 Hz (D5) to 349 Hz (F4).
Flute Note Range: 262 Hz (C4) to 2349 Hz (D7).
Oboe Note Range: 247 Hz (B3) to 349 Hz (F4).
Clarinet Note Range: 147 Hz (D3) to 349 Hz (F4).
Alto Sax Note Range: 147 Hz (D3) to 880 Hz (A5).
Tenor Sax Note Range: 98 Hz (G2) to 698 Hz (F5).
Baritone Sax Note Range: 73 Hz (D2) to 440 Hz (A4).
Bassoon Note Range: 62 Hz (B1) to 587 Hz (D5).

Cut, 0 Hz to 120 Hz (180 Hz), Reduction, Separation.
Between, 120 Hz to 550 Hz, Power, Warmth, Fullness.
Between, 1 KHz to 5 KHz, Honky, Contrast.
Between, 6 KHz to 8 KHz, Rasp, Harmonics, Solo.
Between, 5 KHz to 10 KHz, Shrill.
Roll Off, 12 KHz, Distance, Reduction.

Quality: Fullness at 120 Hz to 240 Hz, shrill at 5 KHz to 10 KHz. Roll off some highs according to distance.

Reduction: For the higher instruments alike trumpets and some trombones, cut a lot from 0 Hz to 180 Hz. As for lower instruments alike Tuba and Horns cut a lot from 0 Hz to 120 Hz. We do not like the brass instruments behind the drummer, so do not roll of too much.

Compression: The trumpet is by far the loudest of the horns, with a large dynamic range that can reach from soft melodies up to stabs and shouts, not so constant overall levels. When dealing with EQ and compression, you'll often deal with the horn section as a single unit (Group). Apply a good amount of compression on peaks, but stay away from really compressing the main parts.

Reverberation: There's something that adds to the excitement of a horn section when you hear it from a distance, when it's interacting with the room. We tend to use a more roomy reverb sound, hall. Reverb and delay work very well with horns.

Orchestral Instruments

Recording: Orchestral instruments are so many, only composition wise and arrangement can help to keep pathways clear.

Panorama: According to stage position.

Frequency Range:

Harp Note Range: 65 Hz (C2) to 2794 Hz (F7).
Harpsichord Note Range: 44 Hz (F1) to 1319 Hz (F6).
Xylophone Note Range: 392 Hz (G4) to 2093 Hz (C7).
Glockenspiel Note Range: 195 Hz (G3) to 349 Hz (F4).
Vibraphone Note Range: 175 Hz (F3) to 1397 Hz (F6).
Timpani Note Range: 73 Hz (D2) to 262 Hz (C4).
Marimba Note Range: 65 Hz (C2) to 2093 Hz (C7).

Quality: Keep track of the Note Ranges.

Reduction: Cut below lowest note.


Record doubled takes. Use low in the mix so it is not obvious. Timing is important, so maybe manually edit the audio. The different doubled takes can differ in tuning and vocal quality, but most of the time does not need to be returned at all Many modern engineers are using auto-tune processing such as Antares Autotune on almost every vocal. We're not saying it's the best thing to do, we are just saying that it is extremely common. Main Vocals are placed at center and upfront, dead in front of all fundamentals and not fundamentals. Maybe have two different copies running at left and right (doubling), still this must result in centered main vocals (avoid swaying around). A good trick is panning duplicates of the vocals left and right. You can invert the right signal for a real dramatic effect. Also pitch shifting left -4 and right 4 can make a more dramatic effect. But however the vocals should align to center always. You can use all kinds of EQ, when the vocals are monophonic (like al single vocals are) use a dynimic EQ that is for monophonic instruments, these EQ's follow the notes and frequency and cut the same amount as other notes and can make the vocals way more steady.

Top End Boost (Highs) is perhaps the easiest and fastest way to make a vocal sound expensive. When using a more affordable microphone, you can simply boost the highs to replicate this characteristic. The best way to do this is with an analogue modelling EQ. Use a high shelf, and start with a 2dB boost at 10kHz. Experiment with the frequency and amount of boost. You can go as low as 6kHz (but keep it subtle) and boost as much as 5dB above 10kHz. Just make sure it doesn’t become too harsh or brittle. When you start boosting the top-end, the vocal can start to sound more sibilant. To counteract this problem, a de’esser can be used. These simple tools are a staple of the vocal mixing process, and required in at least 80% of cases. If you’re recording in a room that’s less than ideal, room resonances can quickly build up. Find these resonances using the boost-and-sweep technique and then remove them with a narrow cut. For a modern sound, the dynamics of vocals need to be super consistent. Every word and syllable should be at roughly the same level. Most of the time, this can’t be achieved with compression alone. Instead, use automation to manually level out the vocal. I prefer to use gain automation to create consistency before the compressor. But regular volume automation works well too. Using a limiter after compression is another great way to control dynamics. You don’t need to be aggressive with it (unless you are going for a heavily compressed sound). Aim for 2dB of gain reduction only on the loudest peaks. As vocalists move between different registers, the tone of their voice can change. For example, when the vocalist moves to a lower register, their voice might start to sound muddy.

Instead of fixing this with EQ and removing the problematic frequencies from the entire performance, you could use multiband compression to control these frequencies only when they become problematic. For any frequency-based problem that only appears on certain words or phrases, use multiband compression rather than EQ. Sometimes EQ alone isn’t enough to enhance the top-end. By applying light saturation, you can create new harmonics and add more excitement. Use a delay for a modern sound, the vocals need to be upfront and in-your-face. Applying reverb to the vocal does the opposite of this, so is undesirable. Instead, use a stereo slapback delay to create a space around the vocal and add some stereo width. Use a low feedback (0-10%) and slightly different times on the left and right sides. I find that delay times between 50-200ms work best.To add more width and depth to the vocal, try adding a subtle stereo plate on the vocal. You don’t want the reverb to be noticeable, as discussed in the previous tip. Instead, bring the wetness up until you notice the reverb, then back it off a touch. Start with the shortest decay time possible and a 60ms pre-delay to give the transients a bit more definition and room to breathe. Another way to give the vocal a bit of depth and shimmer is to apply subtle chorusing. Again, you don’t want the effect to be noticeable. Add a stereo chorus to the vocal and increase the wetness until you notice the effect, then back it off a touch.

Frequency Range:

Vocals Note Range: 82 Hz (E2) to 880 Hz (A5).

Cut, 0 Hz to 100 Hz (120 Hz), Roll Off, Reduction, Separation.
Fullness, 120 Hz.
Male Fundamentals, 100 Hz - 500 Hz, Power, Warmth.
Female Fundamentals, 120 Hz to 800 Hz, Power, Warmth.
Cut, 200 Hz to 400Hz, Clarity.
Boost, 500 Hz, Body.
Boost, 315 Hz to 1 KHz, Telephone sound.
Boost, 800 Hz to 1 KHz, Thicken.
Vowels, 350 Hz to 2 KHz.
Cut, 600 Hz to 3 KHz, Loose Nasal quality.
Consonants, 1.5 KHz to 4 KHz.
Boost, 2.5 KHz to 5 KHz, Definition Presence.
Between, 7 KHz to 10 KHz, Sibilance.
Around, 12 KHz, Sheen.
Around, 10 KHz to 16 KHz, Air.
Between, 16 KHz to 18 KHz, Crisp.



sAY, 600 Hz to 1.2 KHz.
cAt, 500 Hz to 1.2 KHz.
cAr, 600 Hz to 1.3 KHz.
glEE, 200 Hz to 400 Hz.
bId, 300 Hz to 600 Hz.
tOE, 350 Hz to 550 Hz.
cORd, 400 Hz to 700 Hz.
fOOl, 175 Hz to 400 Hz.
cUt, 500 Hz to 1.1Kz.

EQ: If vocals sound tend to close-up, boost some 120 to 350 Hz. Men range 2 KHz and women 3 KHz, with a wide Q-factor (standard for vocal use). The range trough 6 - 8 KHz are sensitive sibilant sounds to 12 KHz. Boost subtle always. Combining a de-esser can help. Even before EQ look at some manual editing. Wideness and openness at 10 to 12 KHz and beyond. Use a quality oversampling EQ on highs. Sometimes a complete vocal track needs to be processed overall, AAMS Auto Audio Mastering System can help with its reference vocal presets.

Quality: Filtering can make a difference for a chorus section (for instance that is mudded, masked). Boost some, 3 KHz to 4 KHz for our hearing to recognize the vocals more naturally and upfront. Boost, 6 KHz to 10 KHz, to sweeten vocals. The higher the frequency you boost the more airy and breathy the result will be (the more quality EQ you will need). Cut, 2 KHz to 3 KHz, apply cut to smoothen a harsh sounding vocal part. Cut around 3 KHz, to remove the hard edge of piercing vocals. When a vocal sounds boxy, then apply some steep EQ 150 Hz to 250 Hz. This will reduce these levels to unboxy (sounds more open). Boost 2 KHz to 3 KHz (5 KHz) with a low Q-factor to ranging from 1 KHz to10 KHz, to adjust speech comprehensibility. Some slight support here is standard, any microphone will muffle the sound a bit, so we must compensate for this at the 2 KHz - 3 KHz range. Main adjustment ranges, fullness at 120 Hz, boominess at 200 Hz to 240 Hz, presence at 5 KHz, sibilance at 7.5 KHz to 10 KHz. You could add a small amount of harmonic distortion or tape emulation effect. A good trick can be running duplicated manipulated copies of the main vocal left and inverted right, this will be heard in stereo, but not in mono.

Reduction: Roll Off < 50 Hz (< 80 Hz downwards steep), cut below this frequency on all vocal tracks. Use a good low-cut from 0 Hz to 120 Hz. This should reduce the effect of any microphone pops. It is common to use a high pass filter (at about 60 to 80 Hz) when recording vocals to eliminate rumble. The better vocals are recorded, the better they can placed inside the mix. Breathers are a question of style, cutting them is common. If you duplicate a track, do not duplicate the breathers. You can edit out all breathers separately on its own track, and then remove all other breathers from the vocals. Syllables and 'Ts' end sounds (rattleling in chorus) can be faded out. Mostly done by manual editing the vocal tracks. No roll off to make the main vocals even more upfront (keeping trebles intact), roll off background vocals though. When a Main Vocal is not sounding, cut 600 Hz out of all other conflicting instruments except drums. When still having problems do a little more cut at 1.2 KHz, solves all problems. Voice is very easy to make flat, sharp or unnatural, think twice before using EQ. The classical way is to record vocal to tape in Dolby mode, and played without Dolby mode. Digital systems Dolby 740, digital up activator.

De-esser: Expanding or Compressing frequencies between 6 KHz to 8 KHz are in the 'sss' and de-esser range. With a band pass filter. A good de-esser is crucial (extreme reduction, but no ‘ lisp’ effect).  You can also edit all 'sss' sounds manually. To make the vocal more open, boost trebles from 10 KHz > (use oversampling EQ) to make them sound upfront. Consider manual editing before using a de-esser.

Tune and Double: Auto tune or tune the vocals. Maybe to original track mixed with the tuned track together, just copy a ghost track and manipulate. You can even use some stereo expansion or widening. When you do not have enough vocals or background vocals, copy them and double / tune / manipulate. Do not widen at copied tracks for main vocals, but you can widen de background vocals with an stereo expander according to panning laws.

Compression: 1176! To make all vocals sit in the mix, we need compression. Mostly compression on vocals can sound loud and hard, it will be fine inside the mix and keeps it upfront. Background choirs can be of many voices, often compressed on a group combined. A fast attack and release, ratio depends on the recording and vocal style. Usually a soft knee compressor. Long attack times for the transients, and release time should be set to the song tempo (shorter then) with little sustain. Vocals now have more presence and charisma, upfront. Start with a ratio of around 4:1 and work upwards for a rockier vocal. Use a fairly fast attack time, release time would normally be around 0.5 sec. A reduction of 12 dB is common for untrained vocalists. Be careful not to over compress, you can always add more later on. A multiband compressor is a good tool for removing unwanted sounds from vocals, use as de-esser for sss sounds, but also for other unwanted frequencies alike pops clicks and some rumble. Also the multibands can be used for different vocal applications. One Band, 0 Hz to 120 Hz, mainly compress for rumble and pops, use a fast attack. Another Band, 3 KHz to 10 KHz, search for de-essing sss sounds. Start with ratio 5:1 to 8:1. Lower the threshold until sss peaks are hit. Another Band, 4 KHz to 8 KHz, can be used for presence with light ratio's 1:5 to 3:1. 

Compressing the room mics can make your rooms sound huge and add a lot to your mix. Some heavy compression can sound quite interesting as long as your not making it too noticeable. Combining this compression to some moderate saturation can make your mixes jump out. Also, some long decaying reverb can sound interesting. Ultimately it makes the room sound bigger and more acoustically pleasing.

Reverb and delay: Using a large reverb on main vocal is not allowed, it is less direct, the singer will sound backed up. Use small room or ambient reverb, be subtle, making the listener not aware of them or noticeable. Combined with bigger rooms and delay, help make the vocals sound fuller without pushing them backwards. Delay, instead of reverb. A delay can make main vocals fuller, without placing them further back on the stage.  The more delay used, you must pay attention to the center placement of the lead vocals. Use the goniometer.

Reverberation: Reverb for the lead vocals tend to be dry, require a high-quality oversampling device to prevent them from being pulled into a cloud of reverberation. You need a small, unobtrusive reverb with attributes similar to a drum booth. Often combined with a delay works well.  That might blur less than a medium reverb. A delay might be far better on main vocals, especially when you need then to be upfront as in most cases. A delay tail on the front vocals, make the vocals appear with more warmth and appear fuller. Without putting the frontal placement into danger mode (panorama). The more the delay is appearing in the mix, the more it will cover the vocals, using ducking (side chain or not) on the first part of the vocals (transients and a little sustain part) can free up and loose some fuzziness. Record vocals dry and you can apply reverberation in style later on. Use a large amount of small room reverbs on the main vocals, instead of using a larger size reverb. Or double the main vocals. Add one track with a small room reverb. Add another with a bigger room trough a delay (1/4 step) and a gate to stay in rhythm (1/4 step). Maybe use a spaced echo. Anyway it is better to not clutter the vocals with on top reverbs and delays after each other (serial). Separate all reverb channels here (parallel), containing dry signal and reverbed signals. Sometimes expand the reverb or delay outwards. For Main Vocals (Single Track or Group) a vocal room, drum booth or small ambient reverb. Bright reverbs can sound exciting, but emphasize sibilance. No pre-delay to set the vocals upfront. Combining with a delay, using a medium reverb might be just too much. Main Vocals - Try to use a Stereo Reverb with delay tail for the main vocals, place the reverb a little hidden. You might solo the reverb and listen to it and find it a bit loud. Within the vocal mix it might be just right, so don’t be scared by this effect. The dry vocals will mask the reverb a bit. To place a choir into the back requires a long reverb, with a bit of pre-delay and damped high ends. The reverb can be set quite high for our ears to accept the 3d spatial information and fight the masking effect. Experiment with a stereo expander in the reverbs return. For Vocals delay can give more depth and placement inside a mix. Use a stereo delay to add small amounts of delay (around 35 ms), watch out for correlation effects.

Delay: Delay can work out better and makes any instrument stay more upfront, as it tends to keep upfront, a reverb will draw more to the back. Delays are more clear and less muddy/fuzzy, so this helps again to make it stand upfront but still have some space. Lead vocals reverb and delay, it’s all about the mix. Create a dry counterweight by doubling the lead, add EQ, compression maybe a short delay, mix it back in. This way the lead vocals are not pushed back too far, but at the same time sound fatter. A little stereo reverb with delay tail for the vocals may work.

Offside Vocals.

Panorama: Sometimes a main vocal singer is accompanied by one or more vocalists. Mostly placed more left and right from the centered main vocals. According to their stage position. As long as you counteract and balance the stereo field, both speakers are playing the same kind of vocal loudness. The background vocals are spread by panning laws, lower voices more in the middle and higher on the outsides. Basically the settings for these accompany vocals are the same as for the main vocals.

Background Vocals or Chorus.

Panorama: The chorus is always arranged that the higher voices are more outside and lower voices more centered according to panning laws. Use a stereo expander for the chorus to widen even more. Also there are effects that can double or harmonize vocals.

Quality: When a vocal sounds boxy, then apply some steep EQ 150 Hz to 250 Hz. This will reduce these levels to unboxy (sounds more open). Boost 2 KHz to 3 KHz (5 KHz) with a low Q-factor to ranging from 1 KHz to 10 KHz, to adjust speech comprehensibility. Some slight support here is standard, any microphone will muffle the sound a bit, so we must compensate for this at the 2 KHz - 3 KHz range. For bigger chorus you can duplicate tracks and use some automatic tuner, pitch or any modeler, thus slightly changing the color of each copy. Chorus can be layered on several tracks, as for recording chorus maybe 4 to 16 (or more) vocals could be used to generate a nice sounding chorus section. The more natural vocals sound the better. Roll of some great deal of highs for distance, set them at the back of the stage.

Reduction: Use a good low-cut or roll off 0 Hz to 120 Hz. To make the chorus more distanced, lower trebles from 10 KHz > to make them sound at the back of the stage (behind the drummer).

Pitch Shifter : A real-time pitch shifter set to shift -4 and 3 panned more left and right, can be used for doubling and creative effects. Also pointing out doubling, harmonizing and special vocal effects alike the vocoder or voice changers.

Reverberation: Backing Vocals are placed toward the rear, a large reverb with some pre-delay and filtered trebles. Reverb can be generously applied here, so the masking effect will stay away or is overpowered by reverb and brings over the 3d spatial information.

Record vocals dry and you can apply reverberation in style later on. For Background Vocals or Choirs (Group) use a Large Reverb with a Pre-delay of about 25ms (Check the snare reverb for starters). Use an EQ to roll off the highs strongly and the reverb sends them all to the back in distance where they belong. High pre-delay for choirs can send them assigned to the back, to the back rows. Try sending the background vocals to a group track. Set a compressor that compresses the loud section, but leaves the quiet ones uncompressed. When this is used to feed a reverb the loud sections will be dryer and the softer sections wetter.

De-esser: Frequencies between 6 KHz to 8 KHz are in the 'sss' and de-esser range. A good de-esser is crucial (extreme reduction, but no ‘ lisp’ effect). You also edit all 'sss' sounds manually.


Static Mix Reference.

Reading up to here, you should have enough information to finish off the Static Mix as a reference for furthermore mixing. Using the dimensions, quality and reduction. As well as finding some stability with separation and togetherness. Unmasking as much as we can to have clear pathways and at the same time save some headroom. Until now we have discussed first the Starter Mix progressed towards a finished Static Mix. Basically called static because after setting up there is no automatic timeline movement of knobs, faders and settings in the timeline of the mix. In the Static mix, we have setup quality, separation (headroom) and the three dimensions (stage plan). We have discussed why it is better to start with dimension 1 and 2 (starter mix) before starting with dimension 3 (static mix)). We would like to finish off dimension 1 and 2 as well as dimension 3 for a good static mix to finish off. Again the Static Mix is our reference point for furthermore mixing purposes, so we need to be sure we have done our very best to get the highest possible result before we progress mixing more dynamical. Now is a great time to just listen, correct until your completely satisfied. Waiting a day and resting our fatigue ears might be a good idea for a last later on re-check. Be 100% sure you have finished a good reference static mix, or else re-check or re-start, before progressing...

Dynamic Mix.

The rest (after finishing off the static mix reference) is dynamic mixing. Dynamic mixing is taken in account events that can happen suddenly or on a timeline throughout the mix, then most likely will return to static reference mixing levels. Mostly digital sequencers offer a lot of automation possibilities. For controlling some outboard equipment alike a control mixer or plug-in, controllers can help to automate easier. Understand that dynamic mixing is influencing all time lined events, even if this is just hitting the mute button while recording and automated mixing. The mute button can be handy for a static mix, we still see them when used automated as dynamic events.


One important thing to know beforehand is that automation should only take place when you are finished setting up the starter mix and static mix. Adjusting Fader, Balance, EQ, Compression, Gate, Limiter, Reverb, Delay as well as setting up routing, the three dimensions for each instrument, leaving some headroom by separation and togetherness as a mix. This can be seen as fiddly jobs that can take hours, it is. Well most of this Starter to Static mixing is technique. Understanding the material you are working on. Be happy with the Starter Mix and Static Mix before entering Dynamic Mixing (Automation). Often starting to soon with dynamic mixing will mean you need to adjust the Static Mix or even adjust the Starter Mix in a repeated kind of fashion. This is basically not allowed, but sometimes necessary for correcting our mistakes. Better nit to make these kind of mistakes. When adjusting the Dynamic Mix, you might get into an endless loop of adjusting. You will notice that you are swapping between both worlds (static and dynamic), constantly making corrections. It is better to first have the starter mix and static mix completely finished off and then go on to dynamic mixing. When you think dynamic mixing is not needed, we all commonly think different. But dynamic mixing can take longer than a starter mix and static mix altogether. When you have spent up to 4 hours for a static mix, expect dynamic mixing will take up to 12 hours.


Automation Events.

Let's begin by getting clear on what we mean by 'effect': an effect is a device that treats the audio in some way, then adds it back to a dry or untreated version of the sound. Echo and reverb are obvious cases, and you can use pitch-shift and pitch modulation in a similar way. 'Processors', by contrast, generally are those devices that change the entire signal and don't add in any of the dry signal. Things like compressors and equalisers fall into this category: as you'll see from the tips and tricks, processors can often be used as effects in their own right, or as part of an effect chain, but until you know exactly what you're doing and what the consequences are likely to be, it is a good idea to stick to these guideline definitions, as they dictate how you can connect the effects into your system. If a device has a Mix control on it that goes from 100 percent wet (effect only) to 100 percent dry (clean only), then you can be pretty sure it is an effect. If it doesn't have a Mix control, and doesn't rely on electronic delays to create its results, then it is probably safe to assume it is a processor. Effects can be connected via insert points, or the effect send and return loop that is included in most consoles and DAWs (Digital Audio Workstations). When effects are used in the send/return loop, their Mix control should be set to 100 percent wet, so you add back only effected sound to the dry sound, which comes directly through the mixer channel.

Processors, on the other hand, comprise an entirely different water heating appliance filled with piscean vertebrates, as they tend not to need any of the dry sound, other than in a few specialist applications. As a rule, processors such as EQ and compression are connected only via track, bus or master insert points — at least until you have the necessary experience to understand why you might want to break the rules once in a while. Having got that off my chest, let's look at some specific effects (we'll look more closely at processors another time).

We can do so much with automation to make a mix stand out more, we will not discuss every aspect, we only discuss a few often used automation tricks:

1. Introducing new sound events, new instruments or new tracks. Automating the instrument fader level can do some good sound information tricks. Let's say you have a Mix running (playing) and at a certain timeline a guitar starts to play solo. New instruments (alike this guitar solo) can be introduced at a louder level at the start of the first introduction (let's say 1 bar in the tempo line or timeline). Then we will reduce the solo guitar instrument to its normal level. Our hearing will accept newly introduced instruments better (spatial information) when the transients at first are a bit louder than normally played. When the solo guitar plays onward automate back to its normal operating level. Normally this louder setting for 1 to 3 seconds will make our hearing accept the solo guitar (reconization). This can be done by setting the Fader of the Solo Guitar (Static Mix Reference) and then automate the louder timeline event parts.

2. Whenever we need more Emphasis on a part of the mix. Sometimes a chorus or verse will be a bit overcrowded, this can make this part of the timeline a bit misunderstood. For instance on the chorus or verse of a song maybe other instruments (not fundamentals alike drums or bass) do not come out clearly. A good trick is to automate the whole drum group, setting a lower level while playing the chorus or verse part. Or maybe automate the main vocals level a bit louder instead while playing the chorus or verse. Also maybe just automate the reverb section of the drum and bass to a lower level setting while playing chorus or verse.

3. Arranged fade-in and fade-out on single instruments or tracks. When you need to fade-in or fade-out a whole mix, leave this for the mastering stage. We do not usually fade when a mix starts or ends. Also we never automate the master track. So what is left is automation fades to make individual instruments or tracks fade-in or fade-out in a certain way. This is purely based on the material and creativeness.

4. Panorama automation. When single spots appear obscure (alike when main vocals clash with the chorus section). Sometimes while playing a mix certain events just seen to clutter, hiding behind other instruments (masking) or any other reason to shortly adjust the panorama. By doing this we can make some time lined panorama events by automation to avoid instruments overcrowding or masking. Sometimes we need to move a reverb or delay for unmasking them for a short while to make then heard better. You can correct this situation only in the timeline part where this occurs, you can just shortly pan one instrument more left or right, and then returning to its static mix position. The use of an automated stereo expander can do a wonderful job, without touching the panorama setting. When for instance the main vocals clash with the chorus vocals, maybe a little automation on the stereo expander of the chorus vocals can do the trick of unmasking. Also automated widening / expanding effect can help.

You can only correct automation by using automation. We always return to the static mix reference level. Whenever you have automated a fader for instance, you can only correct this by automating it back to the original static level. Setting the main vocal fader (as we do in the starter / static mix) does correct the level, but when we have automated this fader already, this setting will be overruled by the automation. Once you start using automation, you can only correct it by automating more timelines or just edit the automation. Stay away of changing the static mix and always use them as reference. Don't use an offset while automating. Maybe now you understand why finishing of a Starter Mix or Static mix first is important as reference, before starting with Dynamic Mixing (Automation). Automation or dynamic mixing can be time consuming and maybe takes three times as much as finishing off a Starter Mix or Static Mix. But can be very rewarding and creative. Take as much time to correct little instances. Do not add more effects until later on. First we try to correct the static mix with automation, before we will add.

 END of Basic Mixing II, goto Basic Mixing III...

Denis van der Velde

AAMS Auto Audio Mastering System


Basic Mixing III

Mixing and Listening, finishing up a completed mix.

After basically starting a mix and finishing a static mix, progression of a dynamic mix, here we will explain some more technique and effects. For furthermore mixing some more creative aspects will be discussed, alike automation and finishing a mix. Don’t start with automation before ready with a coherent static mix, the static mix is your reference for level, pan, dimensions, your stage plan. Each time using automation, keep in mind the reference static mix setting and return to these settings when the automation part has passed. The static mix provides the basement, the floor plan or stage plan, the foundation of a house, and is called static because from the listeners point of view instruments tend to be in the same location with the same amount of level, pan and dimension.


The tempo is a measure for the rhythm, how fast or slow a track is playing. The tempo is mostly set by the drums or the drum player. Drums are used to define tempo and rhythm. The drum player transfers the tempo and rhythm to all other players. Specially in digital mixing or in sequencing the tempo is unattended by beginning users. Mostly set and forget. But in real life the tempo of a playing band or group of instruments will vary. When sequencing or mixing tempo can be of importance to betray the listener in a rhythmic sense, for this we use a timeline. For instance making the chorus slightly more up-tempo can create a sense of stronger listening, what can be of need to make the chorus stand out a little. Varying the tempo up down 5 or -5 BPM by automation on the correct timeline inside the mix, can create a more natural listening feeling. Knowing something about the song, track or mix intensions and knowing a bit about the composition or have the composer in place can be of importance when setting tempo. Tempo can drag (slower) or tighten-up (faster). Also used as a measurement for effects alike delays synchronizing to tempo can be of importance. Especially the rhythmic instruments like percussion drums must be watched according to tempo. It could be handy to know the delay time, when a reverb or delay placed on a snare, how long the snare can die out, until the next snare hit comes. Sometimes in sync with a bar or beat. Avoiding a reverb to overleap the next hit, give rhythmic percussive instruments a short reverb or use a gate after the reverb. Adjust the effect to fade before the next hit, bar or beat starts.

Some calculations can be made beforehand. For instance 12 Bars filled with each 22 Seconds refers to (12* 240/ 22) = 130.9 BPM.

To calculate delay in milliseconds 60000 / BPM = Delay time in ms.

(60 / tempo in BPM) * 1000 ms = Delay time in ms.

(60 / tempo in BPM) * 1000ms * 0.75 = Dotted Note.
(60 / tempo in BPM) * 1000ms * 2 = Half Note.
(60 / tempo in BPM) * 1000ms * 0,666 = Crotched Triplet.

Pitch and length can be adjusted on digital systems without lowering the tempo. The tempo can be adjusted without adjusting the pitch. Or otherwise adjusting the pitch without adjusting the tempo. Some DJ equipment depend on the BPM, pitch and tempo calculations and automated software, that could not be done on a normal vinyl pickup record player. To sync BPM of two recordings is calculation and skill that DJ's did when they did not have digital equipment at that time. Now days a computer (digital system) can do this job, leaving space and time for the DJ to be creative. Midi becomes more important to tempo, as notes are placed in measures and bars of music, controllers do use mostly midi information. The resolution of mid is mostly set at 384 ticks for a single bar of music notation. 192 ticks form half a bar. 96 ticks form a quarter of a bar. (48 = 1/8, 24 = 1/16, 12 = 1/32, 6 = 1/64).

Mix Pix, fight the pix.

For removing pix peaks. Use Stereo Compressor or Brickwall limiter. The peak Pix size is about < 40 Samples long. Use Peak limiters just so that the Pix peaks disappear, visualize with a sample display plugin.

Saturation and Distortion.


Yes, saturation is a pleasing mix colouring tool, but its real genius is its ability to craft texturally interesting sounds that grab the listener.
All of those tubes and transformers in the signal path, not to mention the tape itself, had pronounced effects on the sounds that passed through their circuits.
In particular, the transients those superfast bursts of energy at the start of every dynamic envelope in a sound.
Were shaved down a little at a time by each part of the signal chain, becoming rounder, smoother and softer with every step.
This rounding of the otherwise spikiest, loudest transients is one of the main reasons that sounds which pass through a lot of analogue kit tend to be more well-behaved and quite often easier to mix.
What’s more, in order to make the signals louder than the noise and hiss created by all this gear, the engineers tended to run the levels as hot as possible, which pushed the tubes and transformers to their limits. The result? For every musical note and overtone in the original sound, new notes and overtones rose out of the depths in the form of harmonic distortions. Basic physics dictates that the harder a circuit is pushed, the more prominent the added distortions become, meaning generous amounts of harmonic distortion throughout the mix. We call these added overtones harmonic distortions for a reason, they are harmonically aka musically related to the original notes and overtones within the undistorted sound.
Since this harmonic distortion is inherently musical, people tend to describe its sound in terms like euphonic, rich and warm.
The merging of the two aforementioned effects transient rounding and harmonic distortion adds up to the beautiful phenomenon that is at the heart of my own artistic and sonic life, saturation.
For starters, with all those additional harmonic overtones in play, your EQ literally has more sound to grab within a given swath of frequencies.
So if you’re looking to enhance a vocal’s forwardness in the mix by boosting 1 kHz, there’s going to be more vocal character and personality packed into every decibel boosted, because the sound itself is more dense, vibrant and electric.
Put another way, you won’t just get more vocal presence, you’ll get a more interesting vocal presence. That’s the kind of subtle thing that, when multiplied across 16 or 83 tracks, adds up to something much more compelling than a simple sum of the parts.
Secondly, in dance music, which relies so heavily on the sound of smashed transients for emotional impact (the splatting ‘doosh’ of the snare, the pumping ‘oomph’ of the kick).
Saturation can help solidify sounds. Shaving a fraction of the transient off the front edges of your sounds gives the compressors more room to breathe.
It gives them more controlled signals to work with. This is especially true of percussive sounds. And by that I don’t just means drums, I mean stabbing keys, plucked arpeggios, basslines, exotic stringed instruments.
Any sound that comes on fast and hard, and where the beginning of the sound has far more amplitude than the sustain. Think of it this way: if the sound you’re feeding into a snappy compressor is already smoothed on the front edges, the device won’t have to sweat a bunch of fast impulses that are, relatively speaking, out of proportion to the bulk of the energy they’re trying to shape and control.
Saturating a sound after compression allows you to dial in even more colour. Saturation is a formidable and often underused tool in the mixer’s arsenal. It enhances and even creates new textures in a source, and it allows for a finer degree of control over transients. As such, it can have a powerful effect on the overall clarity and punch of the mix, which in turn can make the difference between a production that merely sits at the edge of the speakers and one that physically jumps out of them.
When you use your tools like that, when you can make texturally interesting sounds and get them to jump out of the speakers and grab someone’s attention.
You’ve gained tremendous ground in the battle for the listener’s hearts and, ultimately, their love for the music.

Reverse Reverb

An enduringly popular effect, with all sorts of uses for vocals, drums, guitars and synths, is genuine reverse reverb. This is where a reverb tail appears to increase in volume ahead of the sound that gives rise to it — a completely unnatural sound and something that's impossible to create in real time. It's easy enough to do in a DAW application though. First, find the section of audio you want to treat and reverse it. In the DAW I use, Digital Performer, there's an offline plug-in for this. Now apply conventional reverb to this 'backwards' audio — either by playing it through a reverb plug-in and recording it to another track, or by rendering it using an offline process (if your DAW offers this). In either case make sure you allow enough additional time at the end of the audio to capture the full, final reverb tail. In the screen grab I've done this by allowing 1500 milliseconds of post-roll processing. Then, reverse the resulting audio. The original audio plays correctly once more, but the reverb you just applied is reversed. You may need to manually realign the audio in your track, to accommodate the extra length of the first reversed reverb tail.

Stereo Expander.

Mostly used in panning law situations when the sound even needs to be more outwards. Check the correlation meter or goniometer. It is better to place widening effects on groups. Never control panning trough groups, only by its individual channel. Never control straight panning or expanding with automation, just small panning and expanding settings for clearing a mix temporarily then setback to the original static mix reference value.


The range of available effects now days is vast, big and versatile. We cannot inform about every available effect over here, but we start off with the most common ones. Some music styles are soly generated by the cause of effects. Some instruments or sounds are soly generated by the cause of effects. Although we have discussed EQ, Compression, Gate, Limiter, Delay and Reverb before, all effects are of importance. An effect behaves by changing the dry input signal. So we could even say fader and panning are effects, but we will not for this mixing example. We have discussed effect placement on single tracks, group tracks and send tracks, as well as pre-fader and post-fader. Panning laws, dimensions, etc. Experience and knowing what can be used and where to place an effect is important. Experiment and learn, learn from others work and experience. Always stay at bay for correlation, masking, separation and togetherness. Use effect to manipulate original sound or to manipulate the dimensions (stage plan) at first. Be creative but try to remember the mixing rules.

Effect Tools.

Tough fader and balance are not really effects; they are tools of a mixer. But EQ, Compression, Gate and Limiter are effect tools we commonly use in mixing. Especially for the purpose of Quality, Reduction and the Three Dimensions (Stage Planning), these tools are commonly used. Some more tools are Dynamics, DE clicker, Denoisers, Expanders, Harmonics and Exciters. When we are looking for separation, togetherness, quality, reduction or stage planning inside the three dimensions, we can first address these tools.

Effects Based On Nature.

Most common are Reverb and Delay. Sometimes Echo, Pitch and Stereo Effects could be used. Basically nature effects have to do with depth and distance (location). For dimension 3, to have any effect on the listener for measuring distance or depth, we need dimension 1 (balance) and dimension 2 (frequency spectrum) in place. Finally when setting dimension 3 with an effect (especially reverberation), makes our stage planning come to life. Pre-delay is an important factor for setting distance, as well as rolling off some high trebles (in dimension 2). For as our hearing perceives the first returns of reverberation of the dry signal and the amount of high trebles as distance. The reverberation by itself is mostly causing our hearing to recognize a room or space.

Effects Based On Artificials.

The biggest group. Some we do explain and discuss over here. Some common examples are Flanger, Phaser, Modulation, Filters, De-Esser, etc. But this group of effects is so vast and versatile, we cannot even name or discuss them all. For instance the Flanger and Phaser are basically originated from reverb or reverberation. But a flanger and phaser use such small time settings, they are normally not produced by nature. Basically when effects can be used for being creative, we place them in this artificial group. Effects that perceive distance or depth can reside in the Effects based on Nature. And effects that can be used as common mixing tools can be placed in the Effect Tools group.

DE clicker.

A DE clicker is used for removing clicks and scratches. More common on single instruments or single tracks. Also mostly used by processing audio offline. As a nasty outcome, the DE clicker by its setting can generate clicks. So watch out. DE clickers can have well to very bad results, only use them when they work. Sometimes it is easier to remove clicks by just cutting them out of the audio (manually). Use a gate to remove long standing clicks.


A denoiser used for removing noise. At first this effect might seem a solution for noisy recordings, but still far better not to use. Do not process anything until you are certain the denoiser removes the right kind of noise and does not take away more in its path. Mute or solo listen, listen to the mix. You might hear that the denoiser is doing its job a little bit too much, and then adjust the denoiser until it only removes the noise you want. If a denoiser does not work, don’t use it. A lot of mixes or commercial recordings contain noise, so don't worry that much. Also background noise can enhance the mix and contains 3d spatial information, better not to delete at all. By using the denoiser on a master track for instance, might remove the depth you are working on for so long to create. So do not use a denoiser on a full mix. For a denoiser it is easy to remove the 3d spatial information that is resting in the background (-60 dB). Just in some cases where the equipment just recorded too much noise, use it. Use a gate to remove long standing noise. Better not to use at all, but resort to better recordings at first, use good noise free equipment instead.

Exciters and Enhancers.

Often used by inserting an enhancer to a group track for effects sending. You could send all instruments or tracks that need to be upfront to the enhancer, while keeping out instruments or tracks that are more distanced. When you need contrast inside a mix, exciters and enhancers make a mix sound better and work best on the complete mix or maybe some groups. Stereo Exciters spread the signal, watch the correlation meter. Use exciters scarcely only when you need to influence the sound, when frequency ranges overleap and the mix is dull. Sometimes used after compression and Denoisers, just to enhance the sound a bit back to its original behavior. Still if your ears are fatigue, do not add any of these exciters or enhancers. Then take a good rest and come back later.


The amount of expansion that is applied is usually expressed as a ratio, such as 2:1, 4:1, etc. While the input is below the threshold, a change in the input level produces a change in the output that is two times, four times, etc, as large. Basically this does the opposite work of a compressor, sometimes referred to as de-compressor. So with a 4:1 expansion ratio (with the input level below the threshold), a dip of 3 dB on the input will produce a drop of 12 dB on the output. When an expander is used with extreme settings where the input and output characteristic becomes almost vertical below the threshold (expansion ratio larger than 10:1), this is often called a noise gate.

Pitch shifters.

A more creative effect in pitch and time. Pitch shifting could be used as echo effect when panned L R. When using two pitch shifters a chorus effect can be heard, hard left and right for an all-round sweetening trick. Some pitch shifters have auto harmony functions for vocals. A Full Stop tape effect can be created, when you turn the offset down gradually. Pitch spirals are a 70's thing, bypassing a delay in front or behind the pitch shifter and maybe do some feedback loop.  Pitch-shifters work by slicing the incoming audio into extremely short sections (typically a few tens of milliseconds long) and then lengthening each section where the pitch is to be decreased, or shortening each section where the pitch is to be increased. Though cross-fading algorithms and other techniques are used to hide the splice points, most pitch-shifters tend to sound grainy or warbly when used to create large amounts of shift (a couple of semitones or more), though they can sound very natural when used to create subtle detuning effects, using shifts of a few cents. A refinement of the system, designed for use with monophonic sources, attempts to synchronise the splicing process with whole numbers of cycles of the input signal, which makes the whole thing sound a lot smoother but, as soon as you present these devices with chords or other complex sounds, the splices again become audible. Though some sophisticated processors combine pitch detection with pitch-shifting, to generate musically correct harmonies in user-defined keys, simple pitch-shifters always change the pitch by the same number of cents or semitones. In musical terms, that means that only the octaves, parallel fourths and fifths are very useful. Other intervals tend to sound discordant, as they don't follow the intervals dictated by typical musical scales. When using subtle detuning to thicken a sound, I suggest trying values of between five and 10 cents and, where possible, adding both positive and negative shifts, to keep the pitch centre correct. Then combine with the dry sound and adjust the level to control the subjective depth of the effect. This is very effective for fattening up guitar solos or backing vocals. By putting a pitch-shifter before a delay, then feeding some of the output of the delay back to the input of the pitch shifter, you can create delays that keep climbing or falling in pitch as they recirculate. Though not always very useful in a musical context, this effect is often used in TV and film dream sequences. Because large pitch-shifts can sound grainy, it is common to combine the effect with the dry signal, rather than using only the 100 percent effected signal, though ultimately this is an artistic rather than technical decision. Though pitch-shifting is an effect, it is easier to control when used via an insert point. However, if you need to use the effect on several tracks in varying amounts, you can use it via a send/return loop, providing the shifter is set to 100 percent wet. That way, you can adjust the effects depth for individual mix channels by using the send control feeding the pitch-shifter.


To reduce the ' Sssssss' sounds from vocals. Commonly in the 4 KHz to 8 KHz range. Use scarcely only when the ' Sssssss' sounds are just too much heard, reduce them only a bit, not a lot. A good de-esser will do a good job, a bad de-esser or settings will do a bad job. Try to remove anything manually before using. Works great on vocals though.


One special effect I used quite a lot in analogue studios, but which is surprisingly tricky to implement in a lot of software sequencers, is where you feed the left and right outputs of an auto-pan effect to two different effects processors. With this setup, the outputs of the two effects can then be mixed together to create a variety of different modulation-style treatments. This patch always worked well in a send-return loop with a pair of phasers, especially if you also EQ'd the two returns wildly differently. The same setup used as an insert could do great things with distortion and ring-modulation processors, and if you were feeling really adventurous, you could fiddle with the panning rate in real time while mixing down.


Filters are commonly used in dance and house music as a creative tool. Years earlier used to make music alive and have some spacy sounds. Filtering (EQ) is still the best way to solve problems and purposes in dimension 2, the frequency range or frequency spectrum. An EQ is reducing or gaining frequencies. A filter will not reduce frequencies but just leave them intact or just cut them out. Specially used for low and high cuts (reduction). There are many kinds of filters, band pass, low pass, Mids, high pass are common. Modern filters have more tricks like synchronization to tempo or a matrix sequencer. Filtering can make a mix jump out, for instance make a difference to a chorus section that is mudded or masked. As a frequency range tool, filtering with a steep high pass filter or low pass filter is a very common heavy EQ technique. For cutting frequencies below 350 Hz (keeping out of the misery area), or 180 Hz, towards 120Hz (Bass range for Base drum and Bass) or 30Hz (pops, low clicks and rumble) a good high pass filter can be used on all kinds of instruments and tracks. Whenever you need a good roll off, filtering might do a better job than just EQ. Also now days EQ's come with filtering elements as well.


Mostly known for distorted guitars. Distortion can help flat sounds, dull sounds and bad sounds. With full sounds distortion is not commonly used, too much harmonics will crowd the frequency range. Distortion makes a sound more dark and can add some warmth. Distortion alike compression can sustain the signal, by this the lower parts are raised more. Harmonics from distortion can be used to replace the frequency spectrum and can make a difference. Technically, distortion is defined as being any change to the original signal other than in level. However, we tend not to think of processes such as EQ and compression as distortion, and the term is more commonly used to describe processes that change the waveform in some radical and often level-dependent way. These include guitar overdrive, fuzz, and simply overdriving analogue circuitry or tape to achieve 'warmth'. In the analogue domain, heavy overdrive distortion is usually created by adding a lot of gain to the signal to provoke deliberate overloading in a specific part of the circuit. Such high levels of gain invariably bring up the level of hum and background noise, so it may be helpful to gate the source. Though overdriving analogue circuitry is the traditional way of creating intentional distortion, we now have many digital simulations, as well as some new and entirely digital sound-mangling algorithms. The most musically satisfying types of distortion tend to be progressive, where the audio waveform becomes more 'squashed' as the level increases. Hard clipping, by contrast, tends to sound harsh. All these types of distortion introduce additional harmonics into the signal, but it is the level and proportion of the added harmonics that creates the character of the sound. Harmonically related distortion can be added at much higher levels than non-harmonically related distortion before the human hearing system recognises it as such, so there is no way to define a percentage of distortion below which audio is acceptable or above which it is unacceptable. The reason that digital distortion has its own character, which most people find less musically pleasant, is because it is not usually harmonically related to the input signal. For example, quantisation distortion, which results from sampling at too low a bit depth, sounds quite ugly, though many dance and industrial music producers have found a use for it, and some plug-ins deliberately introduce it. The use of overdrive distortion as a musical effect probably originated with electric guitar amplifiers, where the less pleasant upper harmonics created by overdriving the amp are filtered out by the limited frequency response of the speaker. If you use a distortion plug-in without following it up with low-pass filtering (or a speaker simulator) in this way, you may hear a lot of raspy high-end that isn't musically useful. This is why electric guitar DI'd via a fuzz box or distortion pedal sounds thin and buzzy unless further processed to remove these high frequencies. The warmth associated with tube equipment and analogue tape is quite subtle when compared with deliberate overdrive effects. As a rule, if you're trying simply to warm up a sound and you can hear the distortion, you should back off it a little, as there's probably too much of it. Adding a little distortion to sounds such as drums, electronic organs, and even vocals, can help them stand out in a mix, and give substance to a sound that's too thin or uneven. Software guitar amp models often sound more convincing if you use external guitar pedals to create overdrive prior to the audio interface.

Vocal Distortion

The hardest part of mixing is getting the vocals to sit properly. There are a lot of tricks you can apply that can help, but I think one of the most useful is to send the vocal to a bus and insert a compressor there, with a high ratio of around 10:1 or more. Set a low threshold, and a medium attack and release, then, in the next slot, load a distortion plug-in with a warmish sound. Use high- and low-pass filters, set to around 100Hz and 5KHz respectively, and mix a small amount back in alongside the lead vocals. You don't need to add much — it should be almost 'subliminal' — but it can really help to fit the vocal in the track.


Overdrive effects such as the use of a fuzz box can be used to produce distorted sounds, such as for imitating robotic voices or to simulate distorted radiotelephone traffic. In science fiction use as a talk box effect, to make a voice sound more robotic or alike transmitted radio signals. For example when to star fighters talk by their radio (com). The most basic overdrive effect involves clipping the signal when its absolute value exceeds a certain threshold. In rock music and related genres, overdrive is a term used to describe the sound of an amplifier running at high volume, usually deliberately, to the point where distortion (clipping) is clearly audible in the output signal. This distortion may range from a slight added growl or edge with some increase in sustain up to a thick distorted fuzzy sound.


The modulator is a lesser used effect, maybe some unattended. Especially for Bass instruments when producing less harmonics or Keyboards / Synths. All modulation will produce fewer harmonic and is a bit uneasy. Even though it is a good effect and can create some nice sounds. Modulation changes the frequency or amplitude of a carrier signal in relation to a pre-defined signal. Ring Modulation is also known as amplitude modulation. The effect made famous by Doctor Who's Daleks and commonly used in sci-fi. In modulation is the process of varying a periodic waveform or tone in order to use that signal to convey a message, in a similar fashion as a musician may modulate the tone from a musical instrument by varying its volume, timing and pitch. Normally a high-frequency sinus waveform is used as carrier signal. The three key parameters of a sine wave are its amplitude (volume), its phase (timing) and its frequency (pitch), all of which can be modified in accordance with a low frequency information signal to obtain the modulated signal. A device that performs modulation is known as a modulator and a device that performs the inverse operation of modulation is known as a de-modulator (sometimes detector or demod). A device that can do both operations is a Modem (short for MOdulate-DEModulate).


Resonators emphasize harmonic frequency content on specified frequencies. A resonator is a device or system that exhibits resonants or resonant behavior, that is, it naturally oscillates at some frequencies with greater amplitude than others. Although it's usage has broadened, the term usually refers to a physical object that oscillates at specific frequencies because it's dimensions are an integral multiple of the wavelength at those frequencies. The oscillations or waves in a resonator can be either electromagnetic or mechanical. Resonators are used to either generate waves of specific frequencies or to select specific frequencies from a signal. Musical instruments use acoustic resonators that produce sound waves or specific tones. Resonation occurs when you play your speakers loud and maybe a door or glass window will vibrate and make noises. Also instruments (or even all world objects) have their own main resonation point, hot spot. Maybe a vocalist can break a glass by singing at a certain frequency, this frequency (when the glass breaks) is the main resonation frequency of the glass or object.


Flanging is caused by the dry signal and a mixed and slightly delayed second signal. The length of the delay is randomized slightly, but is very short < 10 ms. If the delay is too long > 50 ms, the delay gets the lead and starts to generate its own effect (echo, reverb). The flanger and phaser are artificially created, but however are basically reverberation effects. This effect is now done electronically mainly digital, but originally the effect was created by playing the same recording on two synchronized tape players, and then mixing the signals together. As long as the machines were synchronized, the mix would sound more-or-less normal, but if the operator placed his finger on the flanger of one of the instruments, that machine would slow down and it's signal would fall out-of-phase with its partner, producing a phasing effect. Once the operator took his finger off, the instrument would speed up until its tachometer was back in phase with the master, and as this happened, the phasing effect would appear to slide up the frequency spectrum. This phasing up-and-down the register can be performed rhythmically. A Comb filter is mostly some kind of flanger. Flangers make signals a bit fatter, but not as much as its big brother the phaser. Flanging is a time-based effect that occurs when two identical signals are mixed together, but with one signal time-delayed by a small and gradually changing amount, usually smaller than 20 ms (milliseconds). This produces a swept comb filter effect, peaks and notches are produced in the resultant frequency spectrum, related to each other in a linear harmonic series. Varying the time delay causes these to sweep up and down the frequency spectrum. Part of the output signal is usually fed back to the input (looped, feedback), producing a resonance effect which further enhances the intensity of the peaks and troughs. The phase of the feedback signal is sometimes inverted, producing another variation on the flanging sound. Depth (Mix) - Defines the mix between dry and flanged signal. Delay -Defines minimal distance differences between dry and flanged signal. Sweep Depth (Width) - The height of the notches of the flanging signal, the notches are extra frequencies in the frequency range that are created by flanging and the height of the notches. Between 5 to 50ms delay time, mix control 50%, modulation rate between 3 to 8 Hz. For more drama, increase feedback. Feedback Invert sometimes.


The phaser makes use of minimal differences between dry and phased signal, this mainly in form of an all pass EQ Filter. The phaser and flanger are artificially created, but however are basically reverberation effects. By adding the dry and phased signal there is a fase difference created that is clearly ear able. Another way of creating an unusual sound, the signal is split, a portion is filtered with an all pass filter to produce a phase shift, and then the unfiltered and filtered signals are mixed. The phaser effect was originally a simpler implementation of the flanger effect since delays were difficult to implement with analog equipment. Phasers are often used to give a synthesized or electronic effect to natural sounds, such as human speech. The voice of C3PO from Star Wars was created by taking the actor's voice and treating it with a phaser. A phaser is an audio signal processing technique used to filter a signal by creating a series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs is typically modulated so that they vary over time, creating a sweeping effect. For this purpose, Phasers usually include a low frequency oscillator. Depth (Mix) - Defines the volume of the filter output that is added on top of the dry signal. Sweep Depth (Range) -Adjusts the sweep of the filter. Speed and Rate - The speed of the filters adjustable in seconds. Feedback and Regeneration - Negative and positive feedback to make the dry signal more interesting. In reggae often used on Drums, Bass and Guitar (Piano). Between 3 ms to 10ms delay time, mix control set to 50%, modulation rate between 3 Hz to 8 Hz. Feedback Invert sometimes.


Good effect for spatial displacement also moves sounds backwards in the mix. Chorus can make a single instrument sound as multiple or more sweet. Makes the sound more fat a richer. Chorus is a brother of the flanger but only differs in delay. Chorus is basically an artificially created reverberation effect alike flanger and phaser. A delayed signal is added to the original signal with a constant delay. The delay has to be short in order not to be perceived as echo, but above 5 ms to be audible. If the delay is too short, it will destructively interfere with the un-delayed signal and create a flanging effect. Often, the delayed signals will be slightly pitch shifted to more realistically convey the effect of multiple voices. Chorus is a condition in the way people perceive similar sounds coming from multiple sources, is a simulation of this effect created by signal processing equipment to produce this effect. Between 30 to 100ms delay time, mix control 50%, modulation rate between 3 Hz to 8 Hz. Little or no feedback. Increasing feedback creates a rotary speaker effect.

Chorus and Flanging

Chorus and flanging are created in fairly similar ways, the main difference being that chorus doesn't use feedback from the input to the output and generally employs slightly longer delay times. Phasing is similar to both chorus and flanging, but uses much shorter delay times. Feedback may be added to strengthen the swept filter effect it creates. Phasing is far more subtle than flanging and is often used on guitar parts. With chorus, phasing and flanging, the delay time, modulation speed and modulation depth affect the character of the effect very significantly. A generic modulated delay plug-in allows you to create all these effects by simply altering the delay time, feedback, modulation rate and modulation depth parameters. Most of the time, low modulation depths tend to work well for faster LFO speeds (often also referred to as the rate), while deeper modulation works better at slower modulation rates.

Chorus is useful for 'softening' rhythm guitar or synth pad sounds, but it does tend to push sounds further back into the mix, so it should be used with care. Adding more brightness to the sound can help compensate for this effect. Chorus also works well on fretless bass, but tends to sound quite unnatural on vocals. Phasing can be used in a similar way to chorus but, whereas chorus creates the impression of two slightly detuned instruments playing the same part, phasing sounds more like a single sound source being filtered, where the frequencies being 'notched out' vary as the LFO sweeps through its cycle.

Flanging is the strongest of the standard modulation effects. The feedback control increases the depth of the 'comb filtering' produced when a delayed signal is added back to itself. Because it is such a distinctive effect, it is best used sparingly, though it can also be used to process a reverb send to add a more subtle complexity to the reverbed sound.


Vibrato is a musical effect. Vocal and on musical instruments can produce vibrato by a regular pulsating change of pitch, and is used to add expression and vocal-like qualities to instrumental music. Use 2 ms to 15 ms delay time, modulation rate between 2 Hz to 10 Hz.

Doppler Effect.

The Doppler effect, named after Christian Doppler, is the change in frequency and wavelength of a wave as perceived by an observer moving relative to the source of the waves. The total Doppler effect may therefore result from motion of the source or motion of the observer or motion of the medium. Basically the best explanation is, an ambulance passing by with sirens turned on. Not commonly used, but sometime very creatively added or automated.

Pitch Shift.

Similar to pitch correction, this effect shifts a signal up or down in pitch. For example, a signal may be shifted an octave up or down. This is usually applied to the entire signal and not to each note separately. One application of pitch shifting is pitch correction. A musical signal is tuned to the correct pitch using digital signal processing techniques. This effect is commonly used in karaoke machines and is often used to assist pop singers who sing out of tune. It is also used as a creative effect. They are also used to create effects such as increasing the range of an instrument (like pitch shifting a guitar down an octave). Few pitch-shifting algorithms are transparent enough to allow you to transpose anything by more than a couple of semitones without obvious side-effects. If what you're processing is going through an amp modeller, however, you can get away with much more radical changes. You can even do effective swoops and dives in pitch by progressively increasing the amount of pitch-shifting you apply to a note, and pitch changes of an octave or more can sound good, although they probably won't sound natural at these extremes.

Vocal Widening

One of the send effects I most frequently use at mixdown has got to be the classic vocal-widening patch that I always associate with the vintage AMS DMX1580 delay unit. From a mono send a stereo ADT-style effect is created using two pitch-shifting delay lines, panned hard left and right. Normally, I set the first channel to 9ms delay, with a pitch shift of -5 cents, and the other channel to an 11ms delay, with 5 cents of pitch shift. That said, though, I will often tweak the delay times a few milliseconds either way, as this can dramatically alter the effect's tonality.

Time Stretching.

The opposite of pitch shift, that is, the process of changing the speed of an audio signal without affecting its pitch. Pitch scaling or pitch shifting is the reverse, the process of changing the pitch without affecting the speed (tempo). These are more advanced methods used to change speed, pitch, or both at once, as a function of time. These processes are used, for instance, to match the pitches and tempos of two pre-recorded clips for mixing when the clips cannot be re-performed or re-sampled. A drum track could be moderately re-sampled for tempo without adverse effects, but a pitched track could not.

Tuning Effects.

Tuning effects can be used to tune the instrument or single track. Mostly used for tuning guitars or harp, violin, etc. But however can be used on all kinds of instruments, for the purpose of tuning the mix. When all instruments are in-tune, most likely a better and clear mix will arrive. Spend some time tuning your instruments and you will be rewarded by a better frequency spectrum and composition wise a better mix.

Auto Tuning Effects.

Very welcomed effect now days on vocals and all sorts of instruments (correcting the tuning). Also used for creative effects. A good auto tuner will do a good on vocals, especially when designed for vocal use. But for melody instruments alike Bass also commonly used to tune its lower fundamental frequency range. Tuning instruments can also be done by reverting to the synth or sampling device and tune their settings. Often using a tuner, you can sort out the overall tuning beforehand. Recording in tune would be even better. When all instruments are in tune, you will often get better mix in return. For creative aspects there are quite some recordings around with the auto tuner set awkward.

By contrast, tuning (or pitch) correction processors and plug-ins are normally considered processors rather than effects, but they do have creative uses. The idea behind these devices is to monitor the pitch of the incoming signal, then compare it to a user-defined scale, which can be a simple chromatic scale or any combination of notes. Pitch-shifting techniques are then used to nudge the audio to the nearest semitone in the user's scale but, because the amount of pitch-shift required is usually quite small, the result doesn't sound grainy or lumpy, as often happens when large amounts of pitch-shift are generated. Because pitch tracking is used to identify the original pitch, only monophonic signals can be treated. When used with the human voice, it is important that the pitch correction doesn't happen too quickly, otherwise all the natural slurs and vibrato will be stripped out leaving you with a very unnatural and robotic vocal sound. If only a few notes need fixing, consider automating the pitch-corrector's correction speed parameter so that it is normally too slow to have any significant effect, then increase the speed just for the problem sections. This prevents perfectly good audio from being processed unnecessarily. If you stick to a simple chromatic scale (all the semitones), you also run the risk of the pitch correction moving the audio to the wrong note if the singer is more than half a semitone off pitch. A user scale, containing only the desired notes, generally works much better. Some systems also allow you to dictate the correct notes via MIDI. If the song contains sections in different keys or that use different scales, it is often simplest to split the vocal part across several tracks and then use a different pitch-corrector on every track, each one set to the appropriate scale for the section being processed. If your audio track suffers from a lot of spill, or includes chords, the pitch correction may not work correctly. Where spill is loud enough to be audible, you'll hear this being modulated in pitch alongside the wanted part of the audio as it is corrected. As a rule, chords are ignored, so guitar solos, bowed stringed instruments and bass parts (including fretless) can be processed, and only single notes will be corrected. The main creative application for pitch correction is the so-called 'Cher effect', which is achieved by setting the tracking speed as fast as possible to deliberately generate a robotic-sounding result. It's a matter of taste, but for me, this is one effect that has already been done to death!

Tube Amplifier Simulator Effects.

A valve audio amplifier or vacuum tube audio amplifier is used for sound recording, reinforcement or reproduction. Until the invention of solid state devices such as the transistor, all electronic amplification was produced by valve (tube) amplifiers. Whilst solid-state devices prevail in most audio amplifiers today, valve audio amplifiers are still used where their audible characteristics are considered pleasing,. In music performance, especially used on guitar amplifiers. In the case of electric guitar amplifiers a degree of deliberate, often severe, distortion is intentionally added to the sound, and contributes directly to the tone of the guitar, being by itself a major part of the instrument. Sometimes used in music reproduction in high-end audio. Sometimes used for simulating historic equipment. Mostly giving more warmth compared to a transistor, some believe tube amplifiers are better. This is a debated subject. We can use the tube for more warmth and apply as an effect.

Amp Moddeling

One of the most useful features of guitar-amp simulation plug-ins is that they can help mask some quite serious problems with whatever you're putting through them, without necessarily changing it beyond all recognition. I've found that even relatively clean settings can disguise such horrors as clipping on transients to a surprising extent. If you're ever faced with a badly recorded guitar part (even one that's played on an acoustic guitar, or through an amp), try putting it through an amp modeller.pitch-shifting can work well in conjunction with amp simulation, but other ways of editing and processing the raw guitar file before it goes through the amp modeller also yield interesting results. Reverse reverb, resonation, vocoding and Auto-Tune can all produce distinctive effects. Try chopping small sections of guitar out, for an interesting stuttering effect that's nothing like tremolo. A piece of guitar that's been reversed before being fed through an amp modeller sounds quite different to what you get by reversing a guitar part that's already been through an amp, and this technique can be very effective. Likewise, recording three or four separate tracks of single guitar notes and routing them simultaneously through the same guitar amp simulator sounds very different from playing chords. Re-amping a DI'd keyboard or bass can really liven up a sound, but if you don't have access to a nice amp or amp modeller, you can simulate the effect by sending the audio to a bus with a delay plug-in set to a short delay time and with the wet signal set to 100 percent and dry to 0 percent. Then send the bus's output to another bus with a distortion (or better still, a guitar amplifier emulator) plug-in inserted. This simulates the delay you get from miking up a speaker, and if you blend this in with the DI'd sound, it can give the recording a live feel — especially if you use a convolution reverb to add some 'room' ambience. You may also want to roll off the very low and high frequencies to help get rid of that DI'd vibe.

Vocoder Effects.

Create robotic sounds or sparkle the piano. A vocoder (voice and encoder) is a speech analyzer and synthesizer. It was originally developed as a speech coder for telecommunications applications in the 1930s, the idea being to code speech for transmission. Its primary use in this fashion is for secure radio communication, where voice has to be digitized, encrypted and then transmitted on a narrow, voice-bandwidth channel. The vocoder has also been used extensively as an electronic musical instrument. The vocoder is related to, but essentially different from, the computer algorithm known as the phase vocoder. Whereas the vocoder analyzes speech, transforms it into electronically transmitted information, and recreates it, the vocoder generates synthesized speech by means of a console with fifteen touch-sensitive keys and a foot pedal, basically consisting of the second half of the vocoder, but with manual filter controls, needing a highly trained operator. Modern vocoders are more automated and versatile.

Guitar Amp Simulator Effects.

Very welcomed effect for mixing purposes, not also for guitarists. Versatile and comes often with a bunch of presets. Sometimes containing famous guitar players presets. Containing different kinds of guitar amplifiers, speaker setups, delay, echo, reverb, phaser, flanger, etc. A good tool to give a different feel to guitar instruments, but also used on all kinds of instruments, groups and sends. Good tool to revive a dull sound and is very creative.

Loudness Maximizer Effects.

Mostly used for mastering purposes. A combined effect of gaining and compression (limiting) for the purpose of getting the most dynamic out of a mix, without added too much distortion. Better than using only gain or master fader level and a limiter, a loudness maximizer can give more loudness levels. Only sometimes used while mixing, for soft instruments that have almost no level, to give more loudness. This is more a remedy for a bad thing. For mastering purposes the loudness maximizer is last in line, so only used at the end of the mastering stage after mastering EQ and mastering compression, for instance.

Analyzer Effects.

Audio analyzers are really not effects, but to be mentioned. Analyzers are always great tools for visualization. Analyzers can visualize Level, Peak, RMS, Bit, Spectrum, Spectrogram, Scope, Phase or Correlation Meter. Sometimes visualizing can be very helpful depending on the purpose and mixing skill you’re working on. Simple analyzers are peak, vu-meters or red Led’s. RMS is average level. Spectrum and spectrogram are good tools for working inside the frequency spectrum, finding frequencies can be easier with visualization tools instead of listening. A phase or correlation meter checks for mono compatibility.

Midi Controlled Effects.

Often effects can be controlled by midi messages. Especially when using a hardware controller, knobs, faders and buttons can have a function for adjusting and automation. Midi is a standard for transmitting notes, aftertouch and controller information. Pitch bend and modulation controls are common on most midi keyboards. Also controllers used for mixing purposes can be used for effects control. Hardware midi controllers can give an analog feel to digital systems, avoiding using the mouse. Mixing suddenly becomes easier and more precise with an outboard midi controller. Don't forget that you can create audio-style effects purely through MIDI. For example, using a grid-style sequencer, it's very easy to program in echo and delay effects, just by drawing in the repeated notes and then putting a velocity curve over the top to simulate the echoes fading away. By combining this with automated MIDI control of other parameters — reverb send, filter cutoff and resonance, for example — you can alter the timbre of the repeated note and create dubby-sounding, feedback-style delays. They may not always be the first thing you reach for, but the MIDI effects plug-ins that come with most DAW applications like Logic and Cubase often offer something very different from most audio plug-ins. For example, arpeggiators and step sequencers can be great for use in the composition process, and you can use MIDI note to controller data (CC) plug-ins to generate automation data for the parameters of other plug-ins. As they process only MIDI data, and not audio, MIDI plug-ins put very little strain on your computer.


You don't have to create audio effects in your sequencer. For example, I use the Access Virus synth, which features a simple delay effect, with the added bonus that all its parameters are available in the modulation matrix. One favourite trick involves routing velocity to the delay colour parameter. For parts that get brighter with increased velocity, it adds extra animation and bite if the echoes also get brighter. Unusually, the Virus also features four-way audio panning, so you can position an audio signal anywhere between the main stereo outputs and a second pair. If the second pair of outputs is routed to an external effects unit, you can play with the concept of moving a note around in a space, where its position also determines the treatment it gets. More fun can be had by modulating reverb time and colour via an LFO. The same LFO can then be used to control filter cutoff, EQ frequency and maybe wavetable position too (if your Virus is a TI). In this way, timbral changes happen at the same time as effect changes.


Warning Signs

Effects are fun, and can make mixing a more creative process, but it's worth bearing in mind that they won't help in situations where the basic principles of recording have been ignored! Used with care, effects can help turn a good mix into a great one, but they are seldom successful in covering up other problems. It is also very easy to over-use them — sometimes their most valuable control is the bypass button, and it is certainly worth learning to use the basic effects well before throwing lots of complicated tricks at your sound. As long as you let your ears decide what is right, you should be OK, and a little critical listening to your favourite records will give you a feel for what works and what doesn't.

Finishing a mix is a creative aspect.

Starting a Mix is basically setting up a mix using fader, balance, EQ, compressor, gate and limiter to setup for quality and reduction. Static mixing is bringing the dimensions and stage plan into the game. Then adding dimension 3. To add more effects as we showed all effects above, is placing instruments where they belong. You can add effects for making some more quality on individual instruments or tracks. You can add effects for glueing or welding on a group for making the layer sound better. You can add effects on send tracks as well. Depending on your stage planning or to just get some more quality and reduction, remember adding means more overcrowding. So each time you add an effect, understand that your mix is changing each time. Revert back to the dimensions, quality, reduction, separation and togetherness. Do a check and re-checks to stay in the ballpark of mixing. When finally happy with the static sound of a mix, we can use automation to correct some certain timeline parts of a mix. The time it takes to finish a static mix (80%) is just a 1/4 to 1/3 part of finishing the mix. The last 20% will take 4 times more time to finish, the dynamic mix contains automation an all tricks to make the mix sound correct. A static mix is more know-how and experience can take half a day to finish. The dynamic mix only can be started when the static mix stands as a house foundation, and will take a day or two, also this time is improved by experience but also involves more creativity.


Automation is part of the dynamic mix. We can use automation to be creative or correct certain aspects alike masking, 3d spatial information, balance, fader, etc. Endless possibilities exist for automation. Therefore automation can make or break a mix, do spend a great deal of your time on automation. Then at last we listen to the final mix and really are happy. Only then we can finalize the mix.

Introduction Automation events.

One of the first and most forward use of automation is the introduction of new events or instruments. The listener will always be introduced to new instruments, so we automate the first sounding part (maybe a measure or more) with a louder introducing level. This will make the attention of the listener and will recognize the sound, then after this we reduce the introduction level to its basic static mix level again.

Automation of drums.

If you want to combine the dynamics of a well-recorded drum kit with the pumping excitement you get from heavy compression, send either the overheads only or the entire kit to a buss and insert a nice-sounding compressor there. Set the compressor to a high ratio and low threshold and mix in some of this with the song. You may need to adjust the attack and release controls to get the effect you're after, but you don't need to blend in much of the compressed sound to really add punch and weight to a drum track Programmed or sampled drums rarely have natural authentic sound, as recorded drums. The verse, bridge, and chorus, are important parts. The verse automation level is often reduced, to have more dynamics or headroom.

Automation / Muting.

The mute button is a great automation tool. Basically affecting composition, but at least it is not as boring as all instruments playing throughout the whole mix. Experiment with muting drums or instruments, leaving the vocals.

Automation of fade-outs.

All events inside the mix or ending can be automated to be faded in or out. Be careful not to use automation for a complete fade in or out, this can be done after the mastering process.

Automation of Background Vocals, Vocals and Acoustic Guitars.

Apply a low cut filter switch between 80 Hz and 250 - 400 Hz, each time the main vocal and background vocal play together switch to cut more heavy inside low frequency range. When background solo's switch back. When main vocals sing together with acoustic guitars you can apply the same function.


Finalizing the Mix.

As we have explained and discussed, we are mixing in stereo. So outputting the mix as a stereo track is recommended. Maybe we could use a dithering device for resolution purposes. Internally on digital systems, remember what the calculations are based on. For 32 bit float mixing we do not actually need dithering at all. For 24 Bit or 16 Bit integer mixing we do need dithering. When we did use tracks or samples with a 16 Bit or 24/32 Bit integer format, we definitely need dithering. So it is most likely you need dithering when exporting your mix for mastering purposes. For instance when your final product is CD, you need to dither at 16 Bits. Only when you really have mixed everything entirely with 32 Bit Float or maybe even 64 Bit Float operations, you might decide not to use dithering. As we have exported to mix to a stereo track, try not to use normalizing or any other gaining function. This we can do at the mastering stage. Also do not try to clear up the outcome. This kind of cleaning must be done inside the mixing process and then exported again. Only marginal clearing can be done while mastering, so try to bring out the best and cleanest mix you can! Revert to the starter mix, static mix or dynamic mix, but do not try to adjust the outcome of your mix afterwards before the mastering stage.

Some helpful tips.

As you could work completely independent on composing, recording, mixing and even mastering, it is still a good thing to have some people around. Maybe just for the information or their opinion. In the early days of recording music, at least a few people where needed to help just working with all the equipment. So structure and planning was essential as time is money. They needed the best mixing engineer, best recording engineer and a bunch of other people to manage or produce. Now days a single digital computer system could do this work by only a single person. And this is perfectly understandable. A single human being can now days do almost anything by themselves. Even release tracks, songs, collections of clips or entire albums on the internet. Planning, experience and a good deal of time are needed. Other people might think differently about your mix sounding, so let them listen and judge. You will learn from them and their approaches’ to solve a case. In any other way your experience and level can be judged by measuring other people, adding upwards. It is now days possible to do all by yourself, only do when you have the experience. Sometimes searching advice on the internet can help you, on forums you can maybe drop a question. Maybe visit somebody else their studio or see live bands in action can help to improve stage depth in your mixes. Watch people play their instrument and their commitment can help you imagine how this instrument sounds and can be mixed. Maybe then you can imagine what effect can be used inside your mix or to plan the stage. From aged to modern music, planning the stage still is important. And stage planning goes for most music that can be mixed and is a natural approach to our ears. We are better to apply panning and dimension laws, to ease the listener. Listen a mix at very low volumes, when bass drum, snare, bass and melody still sound good and blend in together, the mix is ok and can be surely played at higher volumes as well as lower volumes. Coherent mixing. Do not use a fade in or out during a mix down. Do not cut beginnings and endings, better to have some free (no sound) at start and end of the mix down.

Audio and Midi Latency

Recording Latency - Latency is a very common problem that plagues inexperienced engineers. While recording, it is best to go into your DAW's options and switch the driver system to ASIO (WDM is usually the default), and set your audio interface buffer settings (which are in it's options) to the lowest that your computer will allow. Setting your buffers below 512MB usually give acceptable latency. Set your DAW back to WDM to hear the full resolution sound of your mixes. ASIO gives lower fidelity but is faster.

The plague of MIDI Latency - Having problems with latency when you use MIDI? Go into your DAW's options and switch the driver system to ASIO, set your buffers low and set your audio interface's buffers to below 512MB. If you still experience latency, you may need to lower the buffers further or upgrade your computer.

Buffer settings - Higher buffers will make the recording environment more stable, but with higher latency. Lower buffers will make the enviornment more volitle, but will reduce latency. If you reduce buffers too far, you will get a very weird, obviously choppy sound. Both your audio interface and your DAW will have buffer settings. Switching between the WDM and ASIO driver systems is another option for reducing latency. WDM gives you the full fidelity of the recording while ASIO allows for lower latency, but with lower fidelity.


Using good monitors - The adjustments you make can only be as accurate as the accuracy of your monitoring environment. Imagine painting while wearing foggy glasses, the painting could not possibly be as detailed as it could be with clear glasses. Monitoring is about both high resolution monitors and the acoustic environment. If you cannot afford an acoustician and building modifications, then you will likely want to deaden the room as much as possible. A very practical home studio fix is to move a mirror around the wall of the room and place an Owens and Corning 703 panel ($18) at every place where you can see the monitors in the mirror from the mix position. You may want to cover the panels with fabric to make them look more attractive.

Monitor placement - You will want to place your monitors symmetrically in the room (each monitor should be the same distance from the wall as the other). You will want to make an equilateral triangle with the 2 monitors and the mix position as the end points. Basically, that means that there should be the exact same distance between you, each monitor and the monitors themselves. Monitors will sound brighter the farther away they are from from the wall (Speaker Boundary Interface Response). For instance, if your mixes sound too bright everywhere else that you play them except in your studio, then you can move your studio monitors a little closer to make them more bright to compensate. Therefore, you can move your monitors with this in mind to achieve a better frequency balance.

Two subwoofers - Using a left and right subwoofer will result in more accurate bass adjustments.


Basic Mixing End.

The goal for a good mix is a warm, clear, deep and punchy sound. Where all events are clearly defined, or correspond to the genre and sound good. Examine every event, less is often better. So now we have discussed all parts of the mixing process. We hope our explanation of this process becomes clearer to you and maybe you know why your previous mixes could sound muddy or fuzzy. You know to finish a starter mix and static mix first using quality and reduction. Adding the dimensions according to your stage plan. You know how to separate instruments and tracks as well as welding and glueing to create some togetherness. Apart from the creative aspects of mixing, there are a lot of technical and commonly used set rules that apply. You will notice keeping your mix natural to human hearing and not overcrowding is the way to go. If you do not exactly know what masking means and sound alike, do learn and know exactly. Cutting (reducing, separation, muting, deleting) is the main tool for success, first cut then raise. Also being tidy, taking time to correct things and do it the only way you know how to do so. Understanding that mixing depends more on common sensible rules, then being creative. At start apply the rules more, at end be more creative. Don't go for loudness while mixing, but go for togetherness as well as knowing how to separate. Understanding all of this material explained before in basic mixing I, II and III, should improve your mixing skills and finally generate a well balanced mix.


The next thing on your list should be mastering, as will explain in our mastering tutorial.

End Words.

We hope you have enjoyed this section and explanation over here.

We tend to add information and keep these pages updated, 
so new information could be added over time. Have Fun!

Denis van der Velde

AAMS Auto Audio Mastering System


Mixing Tips

Before you use AAMS Auto Audio Mastering System, Check the Mix! 

There are a number of audio mixing and editting tips that will help you prepare your mixes before using AAMS.
It is important to know how to prepare your mix, so you can get the best sound for your songs!
When quality is at stake, be sure to read this page and spend some time to get your mixes right.


Audio mastering is a process that stands far from mixing, it is the next stage afther mixing and it is the final stage for sound quality. Actually while mixing we do not attend the loudness much, we mix. What everybody is thinking of 'How to get our mix sound loud'! That is what AAMS Mastering stands for, most likely preferred that your mix will become an adequate to commercial radio, CD or MP3 streaming levels, just to fit in correctly. We do not attend the Loudness War, but we need appropiate levels and professional quality. Also when Mastering a Full Album, AAMS Mastering will make the whole Album Sound as an Album. We name it 'the album sound'. So AAMS can do single tracks as well as full albums, and create a good quality professional sound for you. But however, mixing is an important stage before mastering with AAMS starts. So we ask you to attend some time and thought.

Maintain Punchyness - You will want to make sure that your final mixes are punchy. You will want the bass drum, and the overall punchyness to be a little more than you would expect from the final master. If you are thinking that a bass drum punchyness transformation is going to happen in mastering then you are not on the right path. If your bass drum is not punchy enough, revisit #___ about Subtractive EQ and #___ about Low-frequency roll-offs.

Final mixes do not need to compete with final mastered recordings - Due mostly to higher dynamics, final masters usually sound different, and do not need to compete in volume with final commercial masters. This is especially important because if the final mixes are as loud as a commercial master, then the mastering studio cannot use their sweet limiters and compressors to increase the levels in the ways that make mastering magic.

Focus on achieving a good balance - The main goal in your mixes should be to achieve a good frequency balance and a good volume (level) balance between recorded tracks.

Reference CDs are not always as good of an idea as you might thin - Sometimes mixing engineers have a client bring in a reference CD and the goal is to make the mix sound like the commercially released reference CD. That reference CD is almost always a final mastered CD, and chasing after its sound is often like a cat trying to catch a laser dot. After all, the reference CD has been professionally mastered, and you are comparing a final mix to a final master. Trying to achive that "huge" sound you hear on a commercially released master recording while mixing can stop the mastering engineer from being able to help you actually achieve it. Concentrating on getting a good balance is usually the best main goal.

Don't be afraid to do the work - I learned in the military that while shining boots, there are many methods but the most important factor is the time you spend. The same holds true for mixing -- the more time you spend checking this list against the work on your mixes, the better your recordings will sound.

Don't go overboard with effects - Just because you have them doesn't mean you need to over-use them. This is especially true with compression and EQ. A little bit of compression and EQing goes a long way. Reverb can become over-powering, especially if you don't use pre-delay. You must understand how to use the tools that you have, but it is equally important to know when they should not be used.

Good Mics and Good Preamps - Using high quality microphones and preamps can have a serious impact on your recording. If money is not an issue, we recommend George Massenburg preamps, if money is a factor, the grace designs preamps are a very good value.

Check, Check, Double Check!
0. You should do these mix check steps before you plan to use AAMS.

1. Eliminate any noise or pops that may be in each single track. Apply fades or cuts or mutes to spots containing recorded noise, pops or clicks.  

2. Keep Your Mix Clean And Dynamic. Unless there is a specific sound you need, do not put compressing or processing on the master out of the mixing bus. It is best to keep the master buss free of outboard processing or plugins. Dont add any processing to the overall mix, just to individual channels.There should never be a limiter or loudness maximiser set on the master out mix bus!

3. The loudest part in a mix should peak at no more that -3db on the master bus, leaving headroom. It does not matter How Loud your mix sounds at this time, mixing means mixing.

4. Does your mix Work In Mono? As a final reality check, switch the master buss output to mono and make sure that there is no weakening or thinning out of the sound. In any event, do not forget to switch the bussing back to stereo afther this check.

5. Only when a mix is completed and finished off, and your are happy with the overall mixing sound and quality, then the next fase is Aplus Mastering to do their work. 

6. Normalising a track is not necessarily a good idea.

7. Dont add any fades or crossfades, anywhere. Dont fade beginning or end.

8. Do not dither individual mixes.

9. You can output the mix on a stereo, save your mix in Stereo. Use a lossless format! Using digital equipment Wav 32Bit Float Stereo is a good output format.

10. Do not try to output your mix to a mp3 file, this can mean loss of information! If you do want to send in MP3 files, be sure they are of quality, prefer a bitrate higher than > 192kbps, 320kbps is quite good. 

11. Export your mix out of your sequencer or audio setup in a correct and quality unharming format;

12. Finally, always back up your original mixed files!

13. Put all your files of a single mix (the stereo file, reference songs, text documents or pictures or any file that you need to send) in one single directory.

14. Use a packing program like ZIP, RAR, 7z and pack all files in that directory to one single packed file. Name this file correctly, preferably the track number and name of the track.

15. Backup your files! 

Prefer the following audio formats.


- Uncompressed Audio  : Wav, Aiff.
- Lossless Audio  : Flac, WAVpack, Monkey Audio, ALAC.
- Lossy Audio   : MP3, AAC, WMA (> 192 Kbps).

Mastering Stems

Mastering from stems is becoming little by little more common practice. This is where the mix is consolidated into a number of stereo stems subgroups to be submitted individually. Instead of submitting a Stereo output of your mix, you can send the mixing tracks seperately.  For example you might have different tracks for Drums, Bass, Keys, Guitars, Vocal, and Background Vocals. This will give Aplus Mastering more control over the mix and master. If a master from stems is desired, following the same steps listed above is best for each stem. When submitting stems each file track must start at the beginning and must durate though the end, most mixing sequencers will output this way exactly to the sample. Each stem file should be exactly the same length. 

Denis van der Velde

AAMS Auto Audio Mastering System


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