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Hello and welcome to this week's tutorial! Today we're going to talk about an interesting technique used by some famous producer, which can be easily replied on our DAW, and that allows us to apply an effect on a track, mantaining all the clarity and the transient of the original sound.
The Wet/Dry/Wet trick it's a variation of the Parallel Processing system (also known as the New York Trick), but the Parallel processing is used mainly for Compression, while this technique is more suited to effect a track with a Delay or similar effects, and it's best suited for a central guitar (such a Lead Guitar), for a Snare Drum or for Vocals.
How does it work?
We have to create TWO mono FX Tracks, (instead of just one, as it happens with Parallel Processing), and to Pan them as much as you want in the stereo field, one left and one right (that's why we need a central instrument).
Now we must load on these two tracks the same effect, for example a Delay and select the amount of effect to be fed on the fx send of out track on a pretty low level: it has to be heard but it must not completely cover our original sound; Equalize the effect track, if needed, to select the frequency range to affect: usually the area that benefits most of Delay is the one that goes from 150hz to 2000hz, if we effect too much the low end, it's gonna become muddy and resonant.
Now comes into play the reason why we created two tracks instead of one: we're going to apply different settings on the two Delays.
This really goes according to the taste, but if they're tempo synced we can for example switch the rhythm of one of them into a polyrhythm (e.g. 5/4), or we can set one delay on quarter notes and one on eight notes, and so on.
Once we have achieved a thick, interesting double layer of delay, it's time to regulate the right amount of effect to feed to our track: remember that our aim is to keep the original dry sound loud and clear and just to apply, underneath, the effect tail, to give it that cutting through, professional, non-soaked effected feel that can be heard on the best productions.
Hello and welcome to this week's article! This tutorial is to be considered as an expansion of the Autotune article, since it shares with this the same basic concepts, but the Vocoder is a little different: Vocoder is the contraction of the two words "voice encoder", and it consists into an algorhitm that takes a sound (usually a voice) and processes it throug a synth, basically morphing it and tuning it on the note chosen by the player, with a keyboard, or writing it on a piano roll Automation (click here for an article about automations).
This effect has been widely used throughout the '70s and '80 by the first electronic music bands, such as Kraftwerk or Giorgio Moroder, and by some progressive rock band as the Alan Parsons Project, and today the effect is used to give a voice the typical metallic sound of that time.
A Vocoder can be programmed almost the same way an Autotune can be: First we need to create an audio track, which will contain the Vocals we want to effect, then we load into this track's insert the Vst Vocoder. At this point we'll need to create a midi track, choosing the Vocoder as output. Now we can draw on the midi piano roll of this track the notes we want our voice to go through, or we can play them in real time with a midi keyboard.
Here are some cool Vst Vocoder downloadable for free:
Tal Vocoder - an interesting vintage sounding Vocoder
Voctopus - 8 band real time Vocoder with built in synth
Hello and welcome to this week's article! This is the fifth article about modulation effects, and today we're going to talk about ring modulator, octave and harmonizer!
- Ring Modulator: This effect is a sibiling of Tremolo, which, instead of having its amplitude rhitmically altered, it uses a copy of the sound itself very altered, or another sound. Let's make an example: we put on a ring modulator two sounds: the sound of our voice and the sound of a guitar, the effect is like the guitar is "talking" to us, because the effect won't let out the single sounds, but just the "interaction" between them (e.g. Peter Frampton's "Talking Guitar").
- Octave: this one's much easier to explain, it consists into analyzing the input sound and creating a synth copy or more, one octave or more lower. This produces a deeper and more "bassy" sound, like the one that can be heard on Led Zeppelin's song "Fool in The Rain" guitar solo. A particular type of Pitch Shifter that not only works with the octaves, but that alters the sound's pitch widely, even of many octaves, it's the Digitech Whammy pedal, often used by Rage Against the Machine guitarist Tom Morello.
- Harmonizer: this is a very particular type of pitch shifting (sometimes it's called intelligent pitch shifter), in which the effect creates a synth copy of the original sound that respects a certain harmonic distance (for example a third major, or a fifth minor...). This effect is often used to process vocals or guitar, and it can be set at a certain scale related with the original sound, or midi-driven, and in this case it can change its scale at a pre-programmed moment. This effect is part of the trademark sound of many guitar players, from Brian May of Queen to Blind Guardian's guitarist Andrè Olbrich.
Unfortunately there are no common basic controls to analyze for these three effects, except for the Mix control, which sets the amount of original signal to leave unprocessed. Then, according to the type of effect, we will have an octave control for the octave, a key control to set the scale on the harmonizer, and a control that will set the shape of the waveform for the Ring Modulator. Some Octave processors for guitar also feature a Distortion control to add some growl to the processed signal.
Today some DAWs already feature some basic Octave, Harmonizer and Ring Modulation Effect, but if you want to try something new and different here's a selection of the best freeware effects available:
This week we analyze an interesting branch of modulation effects, the one more tied to the "vintage" and psychedelic sound of sixties and seventies: Tremolo, Vibrato and Rotary Speaker.
Tremolo is an effect that simulates a rhythmic opening and closure of the guitar's volume control, it can be set to different timings and it must not be confused with the Tremolo Bridge.
A good example of tremolo can be heard at the beginnig of the Nancy Sinatra's song "Bang Bang".
Vibrato, instead, is an effect that rhythmically alters the pitch of a sound, through a frequency modulation. It is easily achieved with vocals (with the singer periodically contracting its diaphragm) and string instruments, making the string to vibrate with our fingers or the whole bridge to move via the Whammy Bar, but there are also many electronic devices that can produce a similar result, and their "warbling" sound often appeared on Surf Rock or Noise Rock records. A good example of this effect may be found at the beginning of the song "Machine Gun" by Jimi Hendrix.
Rotary - Leslie: this last variation is basically an actual speaker rotating inside a box (created by Donald Leslie, thus the name), and its speed can be controlled the musician.
The result is a phenomenon called Doppler Effect: a frequency change of the perceived wave based on the varying of the position of it source. This device was particularly used with the Hammond Organ, a true protagonist of '60s and '70s music, that can be heard on many Deep Purple songs (for example in "Living Wreck").
These three effects, which are quite similar as a result, have two fundamental controls in common (although by downloading some of the free Vst listed below you'll notice that there are many more parameters available):
Speed (or Rate): controls the frequency of the variation, typically from a maximum (very fast) to a minimum, which may be as slow as one cycle taking several seconds.
Depth (or Intensity): controls the amplitude (in volume or in pitch) of the variation. The minimum depth is often zero (no effect on the sound at all), while the maximum depth does not normally cut the sound off completely at the cycle minimum, but may reduce it by as much as making it inaudible.
Today many DAWs already feature some basic Tremolo, Vibrato and Rotary Effect, but if you want to try something new and different here's a selection of the best freeware effects available:
Hello and welcome to This week's article! Today we're going to talk about Delays and Echos! These effects are also called Time Based Effects, because they take the first sound and repeat it continuously at a certain rate, as if you were talking inside of a cave, with the reflections of your voice coming back to you on a delayed time, summing up one another.
Although the two terms are today exchangeable since they refer to the same thing, usually with "echo" we refer to the natural phenomenon, while the delay is the implementation of that in the audio effect;
in common language many manufacturers calls "echo" a warmer, more vintage sound, and "delay" a crispier, more digital sounding effect.
Delay is an effect born in the '50s by recording a sound on a tape and playing it continuously to simulate an echo (the famous "Tape Echo"), but today it is a very common tool featured on pedalboards and racks of virtually any guitar player in the world; the two guitarists that probably own more of their trademark sound to this effect are Robert Fripp of King Crimson and The Edge, from U2 (which is famous for his riffs that are often exclusively based of the layers of delay).
Delay is the basic modulation effect, from which descends almost all of the other modulations: Chorus, Flanger and Reverb are consequences of applying particular settings to a delay, but there is also another use of a extreme settings applied to this effect: the looper.
A looper is a delay which records long phrases of an instrument, up to several seconds (e.g. a guitar riff) and set them in loop, so that for example a guitar player can play a solo above that base.
The most common and specific uses for a delay are the Doubling Echo, the Ping-Pong delay and the Slapback Echo.
Doubling Echo: Is used to create a sound that feels like the original instrument has been doubled by a second one, and to do this the delay time must be very short: 10 to 50 milliseconds.
Slapback Echo: instead is a longer delay effect (75 to 250 milliseconds), and was often used in the '50s records to underline the main vocals, and later in the '80s it's been used on synths and percussions too, to produce creative rhythmic textures.
Today Delay is often syncronized to the metronome of our Daw, to ensure that its repetitions are perfectly in sync with the song, but if we want to make the repetitions to cut through without being overshadowed by the other rhythmic elements of the mix it's suggestable to use the same metronome but with different beats, for example triplets, or a 5/4 rhythm; the delay will still be "on the click", but the repetitions will be noticed more.
Ping-Pong Delay: a particular type of stereo delay that "bounces" the repetitions (sometimes called "Taps") from left to right, sometimes it has been used by Jimi Hendrix and Brian May from Queen, among the others.
Here are the most common controls found on Delays and Echos:
Length (time) - The delay of the repetition in ms (thousandths of a second)
Length (musical) - The length of the repetition in beats (e.g. 1/8 note)
Feedback - How much of the delay output is fed back into the input of the effect.
Wet/Dry (sometimes these two controls are fused on a single control called "Mix") Wet - The amount of processed signal in the output. Dry - The amount of unprocessed signal in the output.
Lowpass filter - A filter, which filters out from the repetitions the frequencies ABOVE this value.
Hipass filter - A filter, which filters out the frequencies BELOW this value.
Today most DAWs already feature some basic Delay and Echo Effect, but if you want to try something new and different here's a selection of the best freeware Delay and Echo Vst:
Readelay, a one of the most used free delays around, very versatile and professional
This time we continue our research for modulation effects focusing on Phaser and Flangers.
The sonic result of these two effects is pretty much similiar, but the phaser is softer, smoother and less extreme, while the flanger is more "Jet-Like".
The Phaser is the first of the two effects to be invented (around the '60), and consists into splitting the signal in 2 identical ones, and into applying to one of them a filter that alters its phase. By summing the two signals; the obtained effect it's a rhythmic phase cancellation that variates according to the settings, thus creating the "oscillating" sound typical of phaser. The Phaser, unlike the flanger, doesn't work by delaying the second signal and summing it to the first one, therefore the sound is less aggressive. Additionally, the output can be fed back to the input for a more intense phase shifting, creating a resonant effect by emphasizing frequencies between the wave notches.
This effect has been use extensively by many bands, especially the psychedelic rock bands of any time, but the most famous guitar player that has used it on its records is probably Jimi Hendrix.
The Flanger effect is produced by summing two identical signals together, with one signal delayed of a small and gradually changing period, up to some centisecond. This produces a phase cancelling effect that variates according to the variating lenght of the delayed signal, producing an effect called "Comb Filtering", named after the visual effect this process does to the waveform. Variating the time causes the two tracks to sweep up and down the frequency spectrum, and the flanger is the effect dedicated to produce this result. Part of the output signal is sometimes fed back to the input, producing a resonance (sometimes even inverting the phase of the fed back signal) which further enhances the intensity of the effect. One of the most famous guitar players that has used this effect on many songs is Edward Van Halen.
Here are the most common controls found on Flangers and Phasers:
Depth: Sets the depth of the flanger and phaser effect, the width of the oscillation. Higher values equals to deeper oscillations.
Resonance: Sets the amount of resonance. Higher values equals to more effect.
Rate: Sets the cycle speed of the effect: higher values meansa faster effect cycle.
Modulation Phase: With this control the effect can be oriented on the stereo field in different time rates between left and right, in order to give it more ambient. To use this effect live we need a stereophonic amplification.
Today most DAWs already feature some basic Flanger and Phaser Effect, but if you want to try something new and different here's a selection of the best freeware Phaser and Flanger Vst:
Hello and welcome to This week's article! Today we're going to talk about Modulation Effects, focusing on the Chorus Effect. Modulation effects are filters that takes a given signal and creates a copy, with a given delay and pitch modification, to be summed with the original one, creating a wide range of different results. The Chorus effect is an effect that can enrich remarkably a sound: it doubles the sound creating a slighly delayed (usually around 20ms) copy whose delay will not be stable but will keep on variating, oscillating 5ms more and less, plus the copy's pitch is slighly detuned, giving the impression that the copy is (in case of a Vocal track) another person singing along with the first one: not identical, but very similiar, and this effect is used to make the original sound wider and fatter. This effect was particularly used in the mid '80s, for Vocals and guitar solos, in order to add more character and make them sound more metallic.
The Chorus' most common controls are:
Depth: This control sets the depth of the effect: the higher the value is, the deeper will be the oscillations. Basically it's the Chorus' intensity.
Rate: This one sets the effect cycle, which is the speed of the Rhythmic detunings, from very slow (less than a cycle per second), if we set it to 0, to very fast (rarely used), if we crack up the control.
Feedback: This control refers to the process of feeding back a part of the effected sound on the input of the effect itself, therefore we decide how much chorus will be sent a second time through the effecting process: this will lead to a much more effected result, good for creative uses.
Pre-delay: This control lets us choose after how many milliseconds from the original sound the effect will activate. A higher value will make the chorus to intervene after a longer time, allowing the original sound to be heard uneffected before it starts affecting it, while if we set it to 0, the effect will engage immediately.
Level: This one sets the mixing level of the Chorus with the original sound: a higher value will lead to a higher wet-to-dry ratio.
Most of DAWs already features a Chorus effect, but if you want to try some cool vst downloadable for free here's our suggestions:
OrangeChorus: one of the best free Chorus Vst Around, very simple.
Hello and welcome to this week's article! Today we're goint to talk abot How to Mix a Guitar solo! Let's start off assuming that we have already recorded our solo, by Microphoning our amplifier, or using a Guitar Amp Simulator plugins, or even a Hardware Processor. Once that our track is ready for mixing, we must keep in mind that our Lead Guitar track will share the same mid-range frequencies used by our rhythm guitars, so we must find a way to make the solo cut through while remaining "in the mix" at a volume comparable to the other intruments. Usually the idea is to push the lead guitar part a little bit more on the mid-range than its rhythmic counterpart, just enough to make it cut through: we can treat it on a way similiar to how we mix Vocals, since usually a Solo takes the place of the voice in a song, therefore we can usually Pan it to the centre of the soundstage, unless we're after some particulare effect.
On the Equalization side we can usually use a wide Hi Pass filter, taking out everything from 160hz down, then (after the Compression) we can do a gentle boost on the "vocal area", somewhere between 3khz and 5khz. It's a good idea also to cut some db on the rhythm guitars, in the same area we're boosting our solo: this will help us to cut through the mix even better (another Interesting way to cut through it's also to use a different guitar or amplifier than the one we have used for the Rhythm guitar Tracks).
On the Compression side we can use very fast attack and release times (even 10ms or less) and a low ratio, from 2:1 to 10:1 according to the amount of dynamic we want to keep, bearing in mind that the dynamic response it's a very important part of a good guitar solo, so we must not over-squash it: let's use less Compression than the amount we have used on the rhythmic side!
On the Effects chapter, it's important to say that a guitar sound is probably the one that accepts effect processing better, so we can use if we want some very subtle Reverb or some Chorus/doubler to give some thickness, but the most commonly used effect on guitar solos nowadays it's the Delay, which helps the sustain and gives to the sound a pleasant "bounce-back" feel. Some Tube Saturation or Harmonic Excitement can help the sound to cut through the mix even more.
Finally, if after all this processing we feel that our solo isn't emerging from the mix yet, our last resort is to Automate the rhythm guitar tracks to lower their volume a bit (half db or a db will usually be more than enough) in order to clear some headroom, to make our solo soar over them.
Hello and welcome to this week's article! Today we're going to talk about Virtual Console Emulation!
This article talks about a particular kind of plugin set between a Saturation Plugin, an Harmonic Exciter, a Mix Buss Compressor and an Equalizer. Looks complicated? It's actually easier than it seems, in facts, like we've already seen with our Virtual Channel Strips article, it's just all about recreating a sound that feels a bit less digital, and a little more like the vintage albums.
Back in the day, in facts, recordings were done analogically, by processing the sound through huge and expensive consoles that, just by passing the signal through them, used to give to the wave a particular colouration, and this colouration, featured on some classic, timeless album, is still today sought after from many sound engineers.
This "Colouration" of the sound consisted basically in some characteristic of the electronic components used for the console, and at the beginning they were meant to be as transparent and hi-fi as possible, but nevertheless the sound was inevitably modified by passing through them to the point that, once a real hi-fi and true-to-the-source recording has been possible, the engineers felt something was missing.
When the Drive knob is raised, those Virtual Console Emulators basically works halfaway between a Harmonic Exciter and a Saturator, so they add a bit of gain and a sligh compression too, and the hi-pass and low-pass filter tries to set the sound on the coords of the ones created with the virtual consoles.
Some of these plugins works as a Summing processor too: a summing processor is a tool that is used to sum together the tracks, not only by stacking and exporting them on a single file, but
adding a slightly 'bigger' and more professional sound, although this is the source of much debate in the pro audio world, since always more audio engineers are sticking with the "In the box" solution without problems.
Here are the best Console Emulation plugins, ordered by price:
- Terry West's Saturn Console emulation: A Free console emulator for single channels or busses with fixed Hi-Pass and Low Pass filters, a warm gain-driven saturator with a option to use the fine US-pre gain compressor and two meters.
- Sonimus Satson: another good console saturation plugin, at an excellent pricing, with adjustable hi-pass and low-pass filter and gain control.
- SKnote Stripbus: Four types of console emulation, hi pass and low pass filter, VU meters, stereo buss compression and a very low price.
- Waves Nls: Three different console simulations (Neve, SSL, EMI), which will add different tonal colourations to your single instrument or mix buss and work as Summer.
- Slate VCC: another console emulation that features 5 classic console models, and features summing capabilities.
Hello and welcome to this week's article! Today we're going to talk about Noise Gates!
When we record an audio from a souce, a microphone or straight with the jack into the audio interface, we may have some unwanted noise, generated for example by the hum of the pickups (especially the single coil ones), the quality of the cable, and every other ring on the chain that brings the sound into the Daw.
Let's start by saying that there are different types of "noise": Hiss (which are the frequencies that produces sibilance), Hum (which is the low background noise), Clicks and Crackles (which are the snappy sounds sometimes presents when digitizing vynil recordings, or when a source goes into peak, distorting the signal) and the Ess frequencies, produces by the sibilance of the human voice.
The noisegate usually works by selecting the typical frequencies that we want to remove (for example the hiss frequencies, in the case of the DeHisser), and once they occour, heavily compressing them, lowering their level to silence.
So we're talking about plugins that are both Equalizers and Gates (a Gate is a Compressor that works on the opposite way: when a signal is below a certain threshold, instead of boosting it, it brings it down to zero decibel).
Also Gates are used when recording an acoustic drumset, to remove unwanted bleed of other drum parts on a microphone (e.g. to remove the snare sound from the kick microphone).
Many commercial noisegates (like Waves, Sonnox or Izotope) features a "Learn" function: you play a part of the track where only the noise you want to remove is hearing, and the program will remove it from the whole track leaving the other frequencies untouched. This function is featured by some free plugin too, as you can see on the list below.
Focusing on guitar sound, the main issue here is the low background noise, or Hum, generated by the pickups and sometimes by the cable too. We need to clean the sound before entering in the amplifier, or, especially if we use the distorted channel, the noise will be distorted and amplified too, resulting in a strong hum and feedback.
If we are recording an amplifier by Microphoning it, we must use a Hum remover OR just manually cut away the silence parts that feature only noise, retaining the full harmonic richness of the played parts.
If we are recording straight to the interface using Virtual Amp Simulators, instead, we can use a Noisegate BEFORE the virtual amp, and before the eventual virtual Overdrive that boosts the amp.
Many DAWs today features a bundled Noisegate that eliminates the hum, but if your bundled noisegate is not good enough or if it's completely absent, here's a selection of the best free Noisegates:
- GVST GGATE - One of the most used ones, very simple.
- 7AMP NOISE GATE - One knob gate with Learn function: you simple hold your guitar strings for a second, or keep your microphone in silence, allowing the plugin to study noise pattern, then you set filtering level.
- FLOORFISH- A versatile multi-purpose Gate, with Learn function.
- How to use a Noisegate to remove unwanted background noise or microphone bleed: Let's load a noisegate on the track we wish to "De-Noise", and set the Threshold control wery low, so that no signal is below it, and therefore there is no gating.
Now raise the Threshold control until only the hum is gated, and adjust the attack and release control in order to decide how fast the Gate should kick in, and for how long it should be active. It's as simple as that, just remember, when you use it to clean up your guitar tone, to adjust it with your distortion on, or the hum will be almost inaudible.
Hello and welcome to this week's article! Today we're going to talk about Guitar Overdrive Stompboxes! Overdrive is the sligh distortion effect generated, at the beginning unintentionally, from the tube amplifiers used around the fifties/sixties on a live environment: in order to produce a higher volume, they were driven at maximum, and the musicians started to notice that the sound was changing: it wasn't anymore clean and full of headroom like it used to sound at low volume, it was angrier, with more harmonic overtones, more compressed and the hot rodded tubes were producing a roar-like effect on the lower frequencies. This effect became more and more sought after through the years, so the producers started manufactoring tools to add Saturation to the amplifiers without driving their volume to the maximum, therefore preserving the life of the Tubes; these tools consisted basically into adding an Overdrive Channel to the amplifiers, with a higher input level and a sequence of Gain Stages in order to make te signal more and more distorted, and also in creating Overdrive Stompboxes of various kind, in order to push the level before even entering in the amplifier input.
Today Guitar Overdrive stompboxes are used basically for two reasons:
1 - Toadd sparkle and a little grit to a clean channel, especially by blues, jazz players and anyone who is looking for a vintage sound.
2 - To boost an already overdriven sound, in order to make a guitar solo cut more through the mix, or to play with the interaction between the overdrive and the distorted channel of the amplifier. This last method is mainly used by hard rock and heavy metal players: to get in the amplifier input with an already high level sound, but with no distortion, adds to the distorted channel of the amp a pleasant boost effect on the mid-hight frequencies, helping to achieve the "chugging" sound requested by the most modern and extreme metal genres.
Focusing on a Rock-Metal sound, the boost effect is achieved setting the overdrive with the level at maximum, the gain at zero, or anyway very low (this control is often used when trying to drive a clean channel, not when boosting an already distorted channel), and with the tone control around 12 o'clock. You can anyway cut or boost the tone setting in order to bring out more or less the mid frequencies, especially when using a strongly mid-oriented overdrive like the most famous of all stomp boxes: the Ibanez Tube Screamer.
There are many Vst simulations of guitar overdrive stomp boxes, reproducing virtually any overdrive pedal on the market, included the legendary Tube Screamer, and among those, we suggest you to try these free ones, which are impressingly good, especially when used for boosting a virtual amp:
Hello and welcome to this week's tutorial! Today we're going to talk about Harmonic Exciters / Harmonic Enhancers. First off let's explain what we are talking about: harmonics are multiples of the same frequency (x2, x3, x4...), so harmonics of 50hz are 100hz, 150hz, 200hz... We could say that an Harmonic Exciter is a tone shaping tool similiar to an Equalizer that adds or boost multiples of a given frequency or frequency range, in order to make it sparkle more.
Usually Harmonic Exciters are used in the Mastering Phase, similarly and alternatively to Tape Saturation tools, in order to enhance certain frequencies, and are used mainly to excite the lower frequencies, and the highs, but sometimes are used (like the Tube Saturation plugins) on single guitars tracks or some drum part too.
Harmonic Exciters often features different bands control, in order to manually select the amount of processing to assign do the lows and to the highs separately, for example. The best Harmonic Exciters, such as Izotope Ozone, even lets you manually pick the frequencies to process and which band to bypass, instead of giving fixed bands. The interesting thing about this kind of processing is that, in order to enhance the lower harmonics for example, these processors will raise their multiples even in the mids and in part of the highs; this is the main difference from a regual Equalizer, and that is why these tools are used mainly to give sparkle to certain higher frequencies, and some thump to the lows. Beware though, for it is very easy to overdo, and to have a harsh, out of control final result, a good suggestion would be to blend this effect through the Wet/Dry control.
There are different type of Harmonic Excitements, given by Tubes, by Tape, Aural Exciters (a transistor type of processing used in the mid '70s) and other types, and their result is pretty different in terms of eq. The most important thing to remember is just to not exaggerate with the lower frequencies control, and just to give a small sparkle with the highs control, remembering also that these excitements will raise the level of the track, so compensate by lowering the Master Volume. Sometimes there is a Bass Delay control too, which is a short Delay used to thicken the lower frequencies, but it must be used very carefully to not mess up the lower spectrum of your track, so when in doubt, avoid using it.
There are many Free Harmonic Exciters / Enhancers Vst available around, here is a selection of the most used:
CLICK HERE TO READ PART 1/2 OF THIS TUTORIAL! - Now it's time for a Stereo Expander (here you can find many of them for free): this is a useful tool that lets you spread the stereo image of your mix, or just some parts of it. You can choose to spread your mix as much as you want, my only suggestion is to expand only the upper frequencies of your mix leaving the lower end of your spectrum intact, in facts the lows should be as "mono" as possible, in order to stay tight and defined. Otherwise, the mix may get confused. - The last ring of the chain is the Limiter(there are many freeware plugins of this kind too, you can try for example this one, the Betabugs W1 Limiter), which is needed to raise the perceived volume trying to avoid distortion. The effect is called "brickwall", and it basically consist in setting a volume limit (usually -0.10 db) and raise the input gain in order to compensate the -6db of our starting mix with the desired -0.10 db of our final mastered track, so let's raise the input knob (or equivalent, according to the plugin interface), until we get to a fair amount of output volume, but not so high that we lose the dynamics of our mix: we don't want to squeeze and distort all of the surgical job done so far!! We can also use more than one limiter instance: the first one to raise the perceived volume, then an EQ to compensate if the limiting is taking out some lower frequencies (which may happen sometimes), and then another limiter to trim the volume, just remember that a limiter should Always be the final plugin, and the last limiter should be POST FADER (on the Cubase/Nuendo interface this means that should be placed on the last 2 slots of the effects insert).
- Once we are satisfied of all of the processing done, it's time to Dither (Click Here to learn more), and eventually, to Remove Dc Offset. This is the final part of the mixing and mastering phase, and it's needed if you have recorded in a format superior than 16bit and 44khz (for example 24bit and 48khz); it brings your track down to 16bit and 44khz (which is the standard format for the audio cds) trying to apply as little data loss as possible. Almost every DAW has a dithering plugin bundled, but if you don't have one, here is Loser, a freeware one, and here's another, Voxengo R8brain. - Once the track is converted in 16bit and 44khz you're ready to trim it, setting the markers at the beginning and the end of the project, set the eventual fade ins and outs, and export the WAW track, which is ready to rock!! So here is our chain:EQUALIZER -> (REVERB) -> (MULTIBAND COMPRESSOR) -> TAPE SATURATOR / HARMONIC EXCITER ->STEREO EXPANDER -> LIMITER -> DITHER
Keep in mind that this chain is no law, anyways, so feel free to try different combinations for the single effects, or even take some of them off the chain if you think they're unnecessary!
Troubleshooting:
- Make sure you are checking everything with METERS and Frequency Analyzers, since, especially through mastering, these tools such as the Goniometer, are important in order to avoid Phase problems. You can point out such problems and solve them also by switching your master to MONO and checking if there are any areas of the mix where some instrument gets cancelled by others.
- If you feel your sound is "oversquashed", the buss compression, summed with the other compressors you've used on your single tracks used in the mastering phase is too strong. You can try different settings, especially on the mastering compressor, or you can even take it out completely if you think it's useless.
- Cheat: If you feel you need a litte bit more presence or loudness, keep in mind that the ear is most sensitive in the 3-4 kHz range, so use EQ to boost that range by a tiny amount, especially in quiet parts. E.G. If you boost a very small amount (0.5 db) at 3.2 Khz, you can achieve some punch to the overall mix, but be very careful, as it's easy to go from a small boost to an annoying stridency. Even 1 dB of boost may be too much.
After we've seen what is a Reverb and which are the different types of reverb available, it's time to take a look at the most common controls featured on the majority of the Vst reverbs, like the ones suggested on the First Part of our Reverb article.
There are LOTS of controls on the most recent Reverbs, allowing us to tweak any single detail of the effect, but the most important controls that will help us on our mixing phase are basically four:
- LENGHT: often called Decay, or Reverb time, or with other names, is basically the "Tail" of our effect. The longer this value is, the bigger the "room" will be, ranging from a "shower-like" ambience to a cathedral, up to the most psychedelic space effects. This will be the core control of our effect.
- PRE-DELAY: is the amount of time that will pass before the effect will be applied to the signal. I.e.: if the Pre-delay is 1 second long, the effect will begin affecting the audio track one second after it's started. This aspect is crucial to control the Transient of our sound: if the reverb start affecting a sound with a fast Transient such a Snare drum too early, it will smoothen it up and push it towards the background of the mix. Increasing the Pre-Delay value, instead, will give to the attack of our snare drum the time to leave its Transient unaffected, thus mantaining its place in the mix, and will give it a pleasant Reverb tail that will begin-some place in the middle of the wave.
- STEREO SPREAD: controls how the reverb will be reflected on our virtual room: the higher the value, the more the reverb will reflect widely on the soundstage, and it's used usually to spread the Upper-Mid frequencies. This control can be used in Mono tracks as well as in the stereo ones, just beware not to overdo with it, as it can create some weird resonance or Eq-Masking problems with the other tracks.
- EQUALIZATION - FILTERING: a solution for the above mentioned Eq-Masking problem is to Equalize the Reverb. Many producers tend to roll off or just High Pass the region from the Low Mids down, starting from around 300hz or less, to preserve headroom and avoid the Reverb Tail to mask the other instruments, since the Mid Lows, and Lows area is the "Mud Area", so the clearer the sound is, the better, unless we don't want a specifically dark sound.
Now that we have found the right setting for our Reverb (typically starting from a preset and tweaking the four basic parameters we have analyzed), we have created a "virtual room" where to put our single tracks in order to make them sound like the musicians are playing together in the same ambient. So like we've already seen on the First Part of this tutorial the idea is to create an FX track, load the Reverb there, Equalize the track, if we feel like we need it, and send it to the single instruments. Then, via the WET/DRY control of each track, we decide the amount of effect to be sent: for example, a little more on the Vocals and the Toms, a little less for the Snare, even less or none for the Guitars.
There is also a final use for our Reverb: the Mastering Reverb. This is to be used only in rare occasions, when our mix is too Dry and thin, and we need to add it a little bit of fullness or realism. It is used especially on electronic or pop songs, where almost every instrument is sampled, so everything may sound a little bit too artificial and harsh. The settings of the reverb should be low, the Wet/Dry ratio set low as well, and it's a good idea to add a high pass filter around 2000hz to avoid reverbering the vocals sibilance, and a low pass filter from 100hz down. Do some test bypassing the effect and turning it on again, to see if it's really useful, but be careful: it can really ruin the mix if overused!
Today we're going to talk about how to use Reverb when mixing!
Everybody knows what reverb is, it's the persistence of a sound that we create, as it reflects into an ambient. Fewer knows how to use it properly, as in a mix reverb is one of devil's favourite tools to create mud and make details disappear. It could take a whole book just to describe everything about Reverb and its various uses, but today we will see only its function in the mixing phase, which is to let the single tracks to sit better in the mix and to smoothen a bit the Transient, making it a bit less "In Your Face", letting it "breathe" a little more.
First off, there are different kinds of reverb, the most importants of which are:
- Hardware Reverbs: This kind of reverb, such as Plate Reverb or Spring Reverb, was used on the early studios, and are based on the physical reflection of the signal, sent on a resonant ambient and taken back with a magnet or microphone. Today, Spring Reverb, for example, is still featured as a built-in effect on some vintage guitar amplifier, and there are many Vst emulators, such as the free Spring Reverb Type 4. A good example of Spring reverb can be found on many songs played by Jimi Hendrix.
- Studio Reverbs: Those are digital or transistor reverbs, that have been increasingly used since the mid '80, with the advent of digital rack fx processors, and are known for their cleanliness and linearity. Today studio reverbs are still used because they do not colour the original sound too much and are very versatile: they can be used with very low settings too, for example to add some depht and room even in the Mastering Phase, after the Compressors, in the mastering chain. Some good free Vst of this kind are: DxReverb Light, Magnus Ambience, Voxengo OldSkoolVerb, EpicVerb.
- Ambience Reverbs: Those are the reverbs that tries to recreate a real ambient, and are used mainly on single instruments (especially with sampled drums, or guitar Amp Simulators, DI Bass, and all those situations where you don't have a microphoned sound, so there is no natural ambience on your mix), in order to have a more cohesive sound, as if all the instruments were played in the same room. Today, ambience reverbs are often Convolution Impulse reverbs, which are reverbs based on the real response of a reverb captured by a microphone. We have already seen them applied on Guitar Amp Simulators on This Article, but impulses can be successfully used for any instruments. A great free convolution impulse reverb plugin is SIR.
The ideal useof reverb when mixing is on a Fx Track, so we can use a single reverb instance with a sound that will be coherent through all the instruments of our mix: this will have the double positive side of giving to the listener the pleasant feeling that the instruments are played live on the same room at the same time, and will reduce the CPU load, since reverbs are some of the worst "CPU hogs" among all plugins. As we've seen of the Fx Tracks dedicated article, with a reverb loaded on this track we can send the effected signal to the single tracks, as many as we want, using the same effect instance and adjusting the amount of effect to be sent to the single tracks via the WET/DRY control of each track: this way we can decide for example to send more reverb to the Vocals track, and less to the Toms tracks, or to the Snare; 100% Wet means that the track is completely effected, while the Dry percentage is the amount of signal unaffected by the reverb.
Hello and welcome to this week's article! Today we're going to talk about Stereo Disposition on a mix. The moment when we usually decide the position of the single instruments in our mix's spectrum usually happens right after the Project Preparation phase, at the beginning of mixing, in the "Balancing Phase", the phase where, without using any plugin, we just try to move the volume faders up and down in order to find a good balance between the tracks, and take note of the tracks that feels "unstable": those tracks that we cannot just let sit in the mix on a single fader position, and that will require further processing in order to find their place. It's a good idea to listen the mix in mono before starting panning (and check again at the end of the mixing phase too) in order to see if there is any Phase Cancellation, that may go unnoticed once the project is panned, but that can create problems (Click Here to read an article about Phase Analizers).
"To pan" an instrument means to set it on a position in the mix that can range anywhere from 100% left, to 100% right: if we put a track on a position that is (for example) 75% left, means that 75% of the sound will come out from the left speaker, and 25% will come from the right one, thus this sound will result closer to the listener than a 100% left sound. We can play with the disposition in order to help every instrument to sit better in the mix, and give to the listener the feeling to be "surrounded" by the atmosphere of the song, or make him feel like he's in the middle of the rehearsal room, or even to have something that is moving from one side to another of the "soundstage". The creative uses of stereo dispositions are countless, and some of the first pioneers of stereoscopy have been Jimi Hendrix and Pink Floyd, while the first Beatles albums were originally recorded in mono.
Let's start by saying, as already stated on our Stereo Expander article (Click Here to check it out!) that Low frequencies are non-directional, while highs are very directional. As a result, pan low frequency sounds (kick drum, Bass) toward the center of a mix, and higher frequency instruments (shaker, tambourine, cymbals) further out to the left and right.
- Drumset: there are Two "schools": the "Performer Perspective" and the "Concert Perspective". I personally prefer the first one, as the second one is generally used just in certain live recordings: to set the "Performer Perspective", pan Snare and Kick to the centre, the Hi-Hat slightly to the left, the Ride to the right, and set the toms at the disposition you would find while playing the drumset. Finally, set the OverHead Microphones should be set pretty wide, often one full left and one full right. To give the "Concert Perspective", instead, just flip the left and right sides, as if someone else is playing and you're in front of the set, facing it :)
- Vocals: It's a good idea to set the lead vocals on the centre of the soundstage, and so the backing vocals, but if there is more than one backing voice you can set them one sligthly left (like 27%) and one slightly right; just remember that is a good idea to keep them balanced in the mix, unless you are not aiming to some particular effect. You can also try to pan two copies of the lead voice slightly left and right and "slip" one of them forward in the timeline of a few milliseconds: you will obtain a particular "chorus-like" effect that is sometimes heard on some commercial song.
- Guitars: the most common technique is to pan a rhythm guitar to the left (usually 60% to 90%) and one to the right (someone even pan them 100% per side), and if you feel that there's still a void around the centre, you can record a third guitar track and leave it to the centre, or copy the two guitar tracks and paste them to the centre of the spectrum (or a bit off centre, e.g. you can put the left guitar copy at 25% right, and the right guitar copy at 25% left) lowering them of 5-6 db, just to "beef up" the mix. The lead guitars are usually panned to the centre, as long as there is no voice on that part of the song, otherwise we can set them slightly aside, in order to avoid them to interfere with the vocals.
- Keyboards / Orchestrations: It's common use to pan the background pads or violins to the extremes, while treating the leads (whether they are orchestral leads/hits or synth leads) like solo instruments, so put them in the centre, or just slightly off-centre if the middle of the soundstage is too crowded; just a few points (like 5%) can make an instrument to interfere with the others.
- Addictional Awesomeness: there are some effects that lets you play with stereophony, such as the Ping Pong delay (a delay whose repetitions bounces from left to right, and that has to be loaded on a stereo track), but if you want a track, a virtual instrument or just an effect to MOVE inside of a song, you can Automate it (Click Here for an article about Automations!).
Hello and welcome to this week's article! Today we're going to talk about how to correct the pitch of vocals, or any other instrument. These pitch correction tools are usually plugins that takes a wave sound and drag it as it was a midi sample, higher or lower on the piano roll in order to match the correct note you need.
In the last few years the technology has been developed in order to limit the inevitable degradation of a sound that is pitch shifted, especially when applied the vocal parts, which are the ones where the pitch correction is more noticed: today the pitch correction is blended with the introduction of a synth line almost inaudible right below the vocal, in order to help it to sound more "in tune".
Obviously the pitch correction part should be seen as a "last resort", to use only if you notice, before starting the mixing phase, that there are some errors that cannot be recorded again, but that at the same time are too big to not be noticed, and that risk to ruin the song. First off let's say that auto tuning tools works fine only with clean, monophonic sounds: any kind of background noise, or rasp vocals, or distorted sound may be misinterpreted by the algorithm and lead to mistakes. It's also suggestable to process the vocal parts without sibilant letters (s), and the guitar - bass parts where there are no slides-bendings.
There are two types of autotuning tools:
1) Automated Pitch Correctors: These are the kind of tools, such as Antares Autotune, or the free Gsnap and Kerovee, where you can choose the key of the song and other parameters, and let the plugin to automatically slide the wave on the piano roll, in order to match in real time the scale used on the song. It's a good thing not to leave it activated for the whole song, but to Automate it in order to be switched on only when needed, and turned off when unnecessary. It is also possible to create a MIDI track and route the plugin on it in order to manually decide the pitch as it was a regular synth, in facts today the industry standard is using Autotune manually, line by line, in Graphic Mode.
2) Pre - Scanning Pitch Correctors: These plugins, such as Waves Tune and Celemony Melodyne, works by pre-scanning the track and creating a "MIDI-like" version of the track itself inside their interface, so that you can use real-time autotuning (like an automated pitch corrector), or correct the pitch manually without creating an external midi track. Offline working (which means "not in real time") takes more time and efforts, but usually leads to a better result, since it leaves to you the decision of which single syllable (or even part of it) to be processed and which not. With the best Pre - Scanning pitch correctors, such as Melodyne, it's also possible to modify the single notes of a chord of an Acoustic Guitar, for example, just like a Midi file.
The famous/infamous "Cher effect" present on the song "Believe" has been obtained by forcing the vocals through wrong scales, or manually writing notes for the voice in order create that unnatural-sounding "glitch" (obviously in this case it has been an arrangement choice, but sometimes it's just a pure error).
In conclusion, regardless to the kind of plugin you may choose, autotuning should be used with extreme caution, only when impossible to record again, and only for the single parts that really needs it, since often is better to hear a more natural interpretation, although not perfectly in tune, than the "Midi-sounding" voice that sometimes can be heard, even on commercial songs. The risk is to have a completely flattened and lifeless voice, altough perfectly in tune :)
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