MODULATIONS PART 6: VOCODER! (with Free Vst plugins inside!!)

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

Braindoc Lpc Vocoder - a Linear Predicting Coding Vocoder

Vocovee - a real time Vocoder


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HOW TO USE VIRTUAL CONSOLE EMULATIONS (with Free Vst Plugins Inside)

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.

- Acustica Audio Nebula: an impressive virtual console emulator

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HOW TO USE HARMONIC EXCITERS / ENHANCERS (with Free Vst plugins Inside)

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:

Harmonic Enhancer by HgSounds

Exciter, an emulation of the Aphex Aural Exciter

X-Cita, inspired by the BBE Sonic Maximizer

Exciter, by Christian Budde

Antress Modern Exciter

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HOW TO: THE BASIC MASTERING CHAIN (free Vst Plugins included) PART 2/2

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.

CLICK HERE TO READ PART 1/2 OF THIS TUTORIAL!

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HOW TO USE REVERB WHEN MIXING (with free Vst Plugins Included) PART 2/2

CLICK HERE TO READ THE PART 1/2 OF THIS ARTICLE

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!

CLICK HERE TO READ THE PART 1/2 OF THIS ARTICLE

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HOW TO USE REVERB WHEN MIXING (with free Vst Plugins Included) PART 1/2

Hello and welcome to this week's article!

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 use of 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.

CLICK HERE TO READ THE PART 2/2 OF THIS ARTICLE, with the explaination of how to set the Reverb controls properly!!

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FREQUENCY / PHASE ANALYZERS AND METERING (Free Vst Plugins inside)

Metering is a very useful way to visually see what the audio is doing; checking the frequency range of a multi-track is a great way to see if your mix has an overabundance of some frequency or other, and where to make corrections. Using a Spectrum analizer is very important in the mixing phase in order to "assign" to each instrument a "core" range of frequencies they should tend to, in order to clean each channel of the useless frequencies (for example using high pass or low pass filters), avoiding to put too many sounds layered in the same area.

Making your track loud enough without squashing the dynamics, is also a priority in these days of over-compressed masters, so metering is very important in the Mastering Phase too, in order to check the whole  track and eventually apply some last-minute corrections.

When mixing or Mastering, it is crucial to use your ears, but it is also important nowadays to have a visual reference of what you are doing, to see how narrow your equalization cuts are, or to point out in an easier way the changes you are making (for example the Compression you're applying) is going to impact to the whole song or to the single instrument's wave.

Sometimes metering is very useful also to find where the muddyness of a song comes from, for example from a reverb applied to a single instrument, so that we can correct the problem by Eqing the effect.

Especially in the Mastering Phase, Metering comes handy to check out the Dynamic Range: using a Limiter, in facts, there is a strong risk of overcompressing the sound, ruining all the dynamics. Therefore when limiting is a good rule to check the analyzer, in order to keep a Dynamic Range (DR) of 10 to 14 db (Dynamic Range is the difference between the quieter and the louder parts). Squashing the sound too much will result in ruining the pleasure of listening.

As I often say on these articles, the best and most expensive DAWs often features some metering tool too, but if you need some free Vst frequency analyzer, the best around are two made by Voxengo: SPAN and AnSpec. Make sure to check them out!

In the Mastering Phase, another metering tool that can prove to be very useful is the Goniometer.

This type of spectrum analyzer basically works as a Phase meter, which means that it tells you if your track has any phase problems (any sound that has some frequency that "cancels" other frequencies, making them disappear). If the meter stays somewhat in the middle, with no parts creating any weirdness you should be fine, or you can check the Correlation Meter: if the meter is close to all the way to the  "+1 side", you shouldn’t be having any phase problems. If there are phase problems, instead, you should go checking your mix for any instrument and/or effect (e.g. the tail of the reverb) that may contain frequencies that may cancel some other. To discover which parts of your mix has  Phase problems, you must SWITCH YOUR MIX TO MONO, this way you'll be able to point out more clearly which frequencies may cause the phase erasement problems.

Goniometers are plugins that not often are found bundled in DAWs, so, among the many commercial ones, here is a free one: Flux Stereo Tool.

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COMPLETING THE MASTERING: DITHERING AND REMOVE DC OFFSET (with Free Vst Plugins)

This is the final part of our Mastering Tutorial, which takes place after all the dynamics are finely tuned, the last eq coloring has been given, and the overall volume has been raised to the desired level.
You could say you've finished... Wrong! There is still one last thing we need to do before giving our song to the voracious music business.

First off, if you think there is not enough headroom on your master, it may be caused by many things, and one of these things is the DC Offset. DC offset occurs when hardware, such as a sound card, adds DC current to a recorded audio signal. This current results in a recorded waveform that is not centered around the baseline, which is "-infinity", and for some reason takes away headroom.
Most of today DAWs have a "Remove DC Offset" function, that may be used to solve this problem.

Once we checked if we needed to solve this Dc problem, it's time to Dither.
This phase is when you bring your mix down to 16bit / 44khz, which is the audio cd standard, if you have recorded and mixed an audio with higher settings (e.g. 24bit / 48khz).
The first thing to check is to not have any clipping through the whole chain, so check the signal, and  it's a good rule to bypass and turn on again every plugin present on the chain in order to check how they affect the signal and if there is any plugin deactivated or that you need to remove.
Now check the meters at the end of the chain; is the signal clipping? Is your Limiter set too hard? Make sure that there is still a bit of headroom left to not lose completely all the dynamics of your mix.
Now you can apply Dithering, and remember to use it only if you have worked with tracks with a bitrate superior to 16bit / 44khz, to reduce the quality to cd standards, preserving the best headroom and clarity achievable.

Now, once you have exported the track, load it again on your DAW or on another single track editor, like Steinberg Wavelab or Soundforge, and check the last details, such as the beginning and ending markers, check for peaks, and if needed, apply fade ins/fade outs on the track.
Most DAWs already features a Dithering Plugin, but if you need one, try LOSER and VOXENGO R8BRAIN, which are free, and absolutely worth a trying.

Now your song is ready to rock!!

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HOW TO USE THE STEREO EXPANDER / ENHANCER (free Vst Plugins inside)

Hello and welcome to this week's article! Today we're going to talk about Stereo Expanders / enhancers / exciters / wideners.

A stereo expander is a tool used to increase the width of your mix; although you may have already set the pan on your single channels, sometimes the global mix may still sound "thin", not enough open when compared to your reference albums.
The reason is in the Mastering Phase: sometimes in this phase, along with the other processors, many producers add a stereo expander/enhancer, in order to open up the mix even more, and let it "breathe" and surround the listener.

Today the best commercial DAWs already features an in-bundle stereo expander, but here is a selection of the best free ones: UPSTEREO, that can also excite some frequencies to help the Mastering, BRAIN DOC STEREO ENHANCER lets you decide the width by frequency, FLUX features a vector display that shows you graphically the stereo imaging, and the WIDEBUG is a simple and low-cpu consuming widening tool.

What do we need to know about stereo expanders? That only some frequencies of the mix needs to be expanded, while others needs to stay modo: we need to expand the highs, from the 1000hz up nice and wide (just increase the Width setting until the mix is open like your favourite reference album), while going increasingly mono as we go down with the frequencies, until we are completely mono from around 512hz down.
We need to do this way since the larger the wave is (so the lower it is), the less directional it becomes, thus if we make it wide we risks to create phase problems.
The "phase problem" is the fact that certain frequencies may contrast others with the result of deleting them, so we need to use a Spectrum Analyzer if we can, in order to avoid such problems.

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HOW TO USE TAPE SATURATION / TUBE SATURATION (free Vst Plugins inside)

Hello and welcome our weekly post! This time we're going to talk about Tape Saturation and Tube Saturation.
What are they? We're talking about plugins that tries to emulate the response of the hardware devices that were once used to record which added a distinctive tonality to the sound, that has gone lost with the arrival of digital recording.

- Tape Saturation: is a low-level distortion introduced when recording to tape, which used to add a particular Equalization cut to a recording. In the past, sound engineers used to achieve this effect raising the level of the tape recorder, so that when tape is driven in this way and the level meets 0db (or a little bit beyond) the level clips, but in the analog realm this is known as ‘soft clipping’. When digital recording became available, sound engineers tried to reduce the analog coloring and distortion as much as possible in order to achieve the highest fidelity sound available, but then they eventually realized that something was missing, and that part of the beauty of classic recordings was given also by the "lower fidelity" of analog devices. Tape saturation plugins emulates the sound of an audio recorded to tape, and can give you a punchier sound, since they basically add a very slight Compression and distortion, pushing the sound also a bit towards the mid-frequencies area, and works well on single instruments that requires to be brought forward in the mix, but today its main use is in the Mastering Phase, since to add warmth to single instruments we suggest to use the Tube Saturation we'll discuss further in this same article.

A great free plugin that simulates analog saturation/compression and can help you bringing your mix to life, and that can be used instead of a buss Compressor in the Mastering Phase, is FERRIC TDS, which has three main controls: DYNAMICS, that works as a gentle Compressor, SATURATION, that adds extra harmonics, and LIMITING, that controls peak performances. Another nice free and very simple Vst to try is 1-TIME.

- Tube Saturation: Tube saturation plugins instead, emulates the sound of an audio being processed by a tube preamplifier, and are great for adding analogue "fatness" and warmth to recordings, adding harmonics to the sound. Increasing the amount of drive you can obtain some nice distortion too, but this is not their first aim, so in order to add real distortion is probably better to use other dedidcated plugins. Tube saturation plugins are quite easy to use: the DRIVE control increases the digital input gain. Play with the balance between the drive and output gain to control the amount of saturation required, and you'll be able to add a pleasant, slightly mid-focused colour and harmonics to your GUITAR, VOCALS, SYNTH, KICK, SNARE and other instruments. Since these plugins tends to be a bit more invasive than Tape Saturators, I'd suggest to use them on single instruments, and then use a tape saturator for Mastering, but this is not a rule, just follow your ears.

Today many DAWs features an in-bundle tube saturation plugin, but here's a couple of good free Vst to try: RUBY TUBE, Nick Crow TUBE DRIVER, Voxengo TubeAmp and Hotto Vintage Tube Warmer/Maximizer.

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HOW TO USE THE LIMITER (free Vst Plugins included)

Hello and welcome to this week's article! Today we're going to talk about Limiters!
Limiters are basically compressors set with a high ratio (often infinite) and a hard knee, and their aim is to create a volume threshold that cannot be exceeded. Usually this function is needed in the Mastering phase, and it allows you to maximize the impact of your mix and to control the headroom of the sound, reducing the peaks in volume and bringing up the quieter parts (headroom is the dynamic range of a sound before it reaches the distortion level).
Another important use of Limiters comes in the Mixing phase: often you will need to limit certain tracks with a high dynamic range (such as snare drum, guitars, bass, and sometimes vocals), in order to make sure that, although the sound may already be compressed, there will be no chance that any peak will cross the chosen threshold. Each instrument must seat on its place, in order to avoid bad surprises in the Mastering phase! The use of Limiters in the mixing phase is not to maximize the sound or squeeze it, as it may happen in the mastering phase, but just to set the ceiling for the single signals, so don't be aggressive with limiters when mixing: the aim is just to mantain the maximum headroom possible!

During the the Mastering phase, looking at the spectrum of your wave (using Meters, if needed) you will probably notice that there are some high volume peaks (around the 0db area), and others which are a little too low, and you cannot just turn up the volume, since the higher peaks will exceed 0db and distort.
So what we need is to reduce those peaks, thus giving us the opportunity to raise the global volume of the song without surpassing the distortion threshold.

There is a limiter in almost every DAW, but Here is a selection of the best free ones, and among those our suggestion goes to the Yohn W1, a clone of the famous Waves L1 Limiter.

Let's take a look to the basic controls featured on this kind of effect:

- Threshold: controls where the limiter will start to kick in (just like a regular Compressor), eliminating everything above the threshold (in this, limiters are different from compressors, since with comp you can control the amount of the reduction via the Ratio Control). The headroom created by removing those peaks is automatically compensated by raising the quieter sounds. Usually a threshold that goes from -2db to -4db is enough to make the sound punchier, preserving the headroom and without compromising the dynamics, but the most important thing is to always check the meters for clipping and distortions, and whenever they occour, try applying less aggressive settings.

- Ceiling: determines the maximum volume reachable by the limiter. E.g. : if we lower the threshold down to -15db and the ceiling to -6db, the threshold will cut all the peaks above -15db and raise the rest, then the global volume will be raised to -6db. In the mastering phase is suggested to bring the Ceiling Control to around -0.1db, or sometimes even -0.01db.

- Release: it's the less important control here, and most of the times can be left to a default of 100ms. As on the other Compressors, the release control determines the amount of time the sound takes to go from full limiting to no limiting.

It is very important to keep a Dynamic Range (the difference between the quieter and the louder parts) between 14 and 10db, not less, or the sound will result overcompressed and flat, and in order to keep an eye on this we suggest you to use a Metering Tool.
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).

- How to use the Limiter: as we've already seen, the base idea is to reduce the highest peaks in order to clear space to raise the overall level of the whole mix, so we must lower the threshold and the ceiling controls of the same amount (many limiters, such as the Waves L2 have an option to link these 2 controls), while keeping an eye to the "Attenuation" meter: we should lower the 2 controls until we have some attenuation, but not too much, just the peaks of the loudest instruments (which is often the snare drum). When we have found the right spot (too much attenuation means distortion and we don't want it! We just want to use the "unused" room to heighten the overall level!), which is when we have just a very occasional attenuation, we have found the right threshold level. Now we can unlink the Treshold and the Ceiling controls, and raise the Ceiling fader up to -0,1 in order to use all the volume we can before the distortion!

Limiters often have other controls like Attack, an in-built Maximizer, a Stereo Expander and others, but here we've just analized the core functions of limiting. Feel free to experiment!

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HOW TO USE MULTIBAND COMPRESSION (free Vst Plugins included)

Hello! Today we're going one step forward on the exploration of some advaced compression techniques (for the Basics of Compression CLICK HERE).

Multiband compressors are nothing but a serie of compressors linked together. They divide the frequency spectrum down into a few areas/slices, depending on how many bands the compressor has (usually from three to five). On most cases, the spectrum is divided into four categories: lows, low mids, high mids and highs.
You can switch a specific frequency spectrum off and on, and you can also determine the frequency range of a specific band, moving it left and right.
For instance, if you want to compress the bass, squash the low mids, ride the high mids and leave the highest frequencies uncompressed it is possible, and all with just one compressor.
Mastering grade multi band compressors can be indispensable tools for reducing the dynamic range of a group of instruments (or the full mix) without modifying excessively the overall sound.
Using this method means that you can work on some areas of your mix without affecting others. Although you may not get the cohesive effect that single band buss compressors achieve it will certainly blend mix elements together on a more surgical way; of course if you feel the need, you can experiment with a mixture of multi-band and single band buss compressors.

Here is a choice of the best FREE multiband VST compressors available (especially the C3 and the Broadcast), and among the others I would suggest REAXCOMP, which is by far the best around.
Here is the Fruity Multiband Compressor, instead, on a page that shows a detailed explaination of the most common controls featured on this kind of plugins.

Here is instead a short example of settings for a 5 band Multiband Compressor:

LOW - tighten up bottom end.
Frequency Range: 0Hz-150Hz
Ratio: 2.5:1
Attack: 20ms
Release: 150ms
Threshold: very low to almost always trigger compression.
Gain: make up gain lost in compression.

LOW MID - tighten up the mix.
Frequency Range: 150Hz-600Hz
Ratio: 3:1
Attack: 20ms
Release: 150ms
Threshold: trigger regularly, but be about 2dB below the point of
rarely triggering.
Gain: make up for compression, or just a little more for warmth.

MID - add punch to the mix.
Frequency Range: 600Hz-1.5Hz
Ratio: 6:1
Attack: 10ms
Release: 150ms
Threshold: set fairly low to almost always trigger compression.
Gain: add 4-6dB or more to make up lost gain and add guts.

MID HI - add presence and increased clarity of individual instruments.
Frequency Range: 1.5KHz-6Hz
Ratio: 3:1
Attack: 10ms
Release: 150ms
Threshold: trigger regularly, but be about 2dB below the point of
rarely triggering.
Gain: add 1-3dB for presence/clarity.

HI - reduce harshness without losing sparkle
Frequency Range: 6KHz-15Hz
Ratio: 2:1
Attack: 10ms
Release: 150ms
Threshold: only trigger when harshness present.
Gain: maybe add 1-2dB to recover sparkle lost in compression.

Gain after each band of compression can be used to shape the sound.

(source:  http://www.dogsonacid.com/showthread.php?threadid=25008 ).

And here is the setting that the world class producer Andy Sneap suggest to use on Heavy Metal guitars, with the Waves C4 Multiband Compressor:  Bypass all bands except one that goes from 65hz to 281hz,
Gain: +0.3
Range: -8.0
Attack: 16.03
Release: 25
Threshold: -26.5

This setting helps taming the lows in the "mud area".

Multiband compression is used much more in mastering than mixing. Since we are dealing with such a wide blend of sources on a master track, some mastering engineers use multiband compression to control certain aspects of a mix, like only tightening up the low end for a punchier bass sound, but multiband compressors can be also found used for drumkits, or vocals.

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HOW TO: THE BASIC MASTERING CHAIN (free Vst Plugins included) PART 1/2

CLICK HERE TO READ PART 2/2 OF THIS TUTORIAL!!

Tutorial version 2.0, august 2012

Hello! Today we're gonna see how to master a song, trying to analize the single steps in order to give to your (already mixed) track the final boost it needs to be loud and sparkling, enough to be compared to the commercial tracks.
Let's start from the assumption that the perfect mix just needs to have its volume raised to 0db on the mastering phase, but more than often other processes are needed in order to achieve a good final result.
Obviously there are many ways to build a mastering chain, as always I'm gonna explain you the basics and suggest some free plugins, then you will adapt these ideas to your project and your plugins.

- First off, load the track containing your mix on a stereo track, on a new project, making sure it isn't too loud (I would recommend to keep the mix it around -6db, to give you enough headroom to work in the mastering phase), in order to avoid clipping; it is also importanto to make sure the track is cleaned of all hiss, pop, crackle and noise before stating to master it, or the problems will get worse.


- The first plugin to add to our buss is an equalizer (for example ReaEq), if you feel there's some frequency to correct; that way you'll be able to shape (very lightly) some general frequencies that doesn't satisfy you completely after the mixing phase, usually eq is used in mastering in order to scoop very lightly the mids (more or less -1db around 300hz), and boost gently lows and highs (+1db around 50hz and 5000hz). After these adjustments, using the same eq plugin, create a high pass filter to remove all of the frequencies under the 30hz, in order to clean the mix of the almost inaudible and useless frequencies, like rumble and breath. 

- If we feel that the final sound is still a little bit thin and dry, and we didn't use a lot of Reverb during the Mixing phase, we can also try adding some Mastering Reverb. It consists into adding a Reverb with very low settings, a regular "Room Size", a low Wet/Dry ratio and a Low Pass and High Pass filter between 100 and around 2000hz. It's very important to not overdo, though, since this can really screw up everything :) The position in the Chain is Typically between the Equalizer and the Compressor, but it can be moved after the Compressor if we feel that the Comp is making the effect too strong.

- Now we must focus on the different areas of the mix, and try to point out if there are certain parts (for example, the drum snare) which are too low, or others (for example, the cymbals) too loud, and try to correct the problem; you can do it with the eq, like on the last point, or use this other method, that is less invasive and "coloring", which is the multiband compression (Click Here for an in-depht article on the topic). 
There are many multiband compressors around, and there are some bundled in almost every DAW on the market, but if you need a freeware one, here is a list where you can choose from. Using a multiband compressor lets you choose graphically which part of the curve (i.e. only the highs on a certain frequency) to compress, leaving the other frequencies of the mix unprocessed, and it's a very useful tool to make aimed corrections.

- Now we can add to our mastering buss a Tape Saturation plugin, that produces a slight Compression and Saturation without squashing too much the overall sound, but this depends mainly by the song and by how much compression you have already applied on the single instruments and on the Mixing Buss. Alternatively we can use an Harmonic Exciter, in order to give some sparkle to the high frequencies, and some thump to the lows.

CLICK HERE TO READ PART 2/2 OF THIS TUTORIAL!!

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