Hello and welcome to this week's article!
Today we're starting a new serie called "How to improve", in which we will add some element to our basic mixing articles.
This time we're talking about 5 ways to improve our amp sim guitar tone, which even if we are using some high end plugin, still needs some polish before it can perfectly sit in the mix.
These tips are to be applied after recording, editing, finding the right guitar amp sim and tweaking it.
1) Impulse responses: impulse responses are the single most revolutionary thing happened to the world of guitar tones in decades, since they can change radically a sound and make it realistic in a way that few years ago was unthinkable without microphoning a real amp.
Get yourself a nice collection of impulses, at least a couple of hundreds, it doesn't matter whether free or paid, just make sure to have enough variety to cycle through them until you find the one (or the combination of more than one) that fits perfectly the song, and in order to do it, use a referencing track with a sound similar to what you have in mind.
A good starting point would be our free IR Pack Trident, which has 14 impulses ready to use.
2) Create a guitar bus: route all the guitar tracks that should have similar tones (e.g. the rhythm guitar tracks) into a stereo guitar bus, in order to control with one fader the general presence of guitars in the song.
3) Compress the minimum necessary: the general rule here is "the more the gain = the more the natural compression", so if we have clean guitars we will need to compress more, while if we have high gain ones, we're going just to use a multiband compressor in the low end area, shaving off a couple of db when needed in the area from 80 to 350hz (usually during palm muting).
4) Equalize with wisdom: here the general rule is "boost wide, cut narrow". We could spend hours talking about how to clean a guitar tone, but in this article let's just add the fact that we need to find balance: the sound must be natural, similar as if it's coming out of a real amp, so:
- high pass up to 50 to 80hz
- if the guitar still sounds weak and lacks of presence, add some body in the area between 1000 and 2000hz, finding where it suits, with a wide bell boost of 1 or 2 db.
- cut a couple of db with a narrow bell in the 4k area, sweeping the eq until you find the "nails on a fingerboard area"
- add some air, with a wide bell around 8-10khz
5) Don't be afraid of saturation: this is another tool to be used if our tone (especially if clean or overdriven) is too weak, thin, and we can't solve by using the eq; a touch of tape or tube saturation can add the weight and the midrange our tone needs without altering too much the sound.
This tool can be used on other instruments too!
Hello and welcome to this week's article!
This time we're going to embark to a pretty huge topic that will touch many of the article already present on this blog, but with this I'm going to try and make order to what I've learned in the past few years and share some extra tip with you.
In order to obtain the maximum from this serie of articles please follow the hyperlinks, they all lead to in depth articles dedicated to the various topics.
The mixing process of an album is almost a philosophical project: it consists into gathering together a puzzle with a million pieces (songwriting, recordings of all instruments...) that would have no meaning on their own and carefully combine them in order to make a vision that until now is only in the head of the artist into a form that anyone can enjoy.
The process can be long and complicated, and it requires first of all order in our head: to help making order we have broken it down into one chapter for each instrument, in order to cover as many topics as possible.
Even before starting putting together our project it's important that we have clear in mind the focus of our mix (click here for a dedicated article): this will be the north start that will guide us through the mix, and will give us a direction towards where the overall sound should aim.
Let's start by assuming that we have a good song, and that this song has been recorded properly: we start with the project preparation phase (click here for a very detailed article), and once we have all the tracks in order, the project tidy and color coded, we make sure that the gain staging is correctly set to make sure that no track (nor the master fader) peaks above -12db.
I suggest to start with a mix bus compressor (click here for a dedicated article) in the master bus set very gently (2:1 to 4:1 ratio, the more dynamic and aggressive the genre, the higher the ratio), and lower the threshold knob until you see you are shaving off not more than a couple of db of peak: this move seems useless at the beginning, but it will add a little of energy to the mix, will help the instruments to glue better together and will allow us to push less with the compressors in the individual tracks, which is usually more noticeable and can be less pleasant.
Moving on with the mix we'll be introducing also all the other instruments, and gradually the general volume will be increasing: we need to lower the faders of the individual instruments in order to make sure that we we'll be still shaving off not more than 2db with the M.B.C., and not peak above -12.
This will be a good way to keep under control the gain staging and the headroom, we're going to need it in the mastering phase.
The general rule is that a mix is made of many 1% moves that all must go towards the direction of achieving an euphonic final result: you can try and experiment with all the crazy plugins and processors you want, but after each move you must always a/b with the plugin on and off (using a reference track at every step to see whether we are going in the right direction): if the sound gets better, leave it, if it doesn't add anything or it makes it worse, remove it. Sometimes less is more.
Now let's start with the drums, the pulsating heart of our song: if we get this right, we are already halfway there. Are we using a drum sampler? Is the drum sampler featuring pre-processed sounds?
In this case we need to spend most of our attention to which samples to use: carefully go through you whole library of drum samplers and individual samples to find the perfect snare, the perfect kick and so on, because once they are there, there is very little room for mixing, since the samples are already processed.
You can find many free and paid drum samplers on this blog by simply typing drum sampler in the search engine.
Once we have our ideal drum sound we need to work with the integrated mixer of our drum samper to make the levels, and we could probably add some high-passed reverb through an fx track (we can high pass the reverb to make sure it affects only from a certain part of our sound up) from 200hz to 1000hz up (this way we will avoid the sample to get too dark and boomy), and to carefully introduce some of it into the snare, toms and cymbal tracks to make them feel like they are being played in the same room.
Usually with pre-processed samples there is no need for extra eq and compression.
This is a good moment to hear the aid that the bus compressor is giving us: try to playback the drum track and turn it on and off from the master bus: you will notice that there is an 1% increase of fatness and roundness to the sound, and this is just one of the many 1% a mix is made of.
If the sound instead starts pumping (meaning that when a loud sound like a snare is played the other ones lowers in volume) it means we need to lower the compressor.
Are we instead using real drum sounds or non processed samples?
Then we need to follow the steps of our in depth drum mixing guide.
Today we are going to expand our knowledge about Limiters by clarifying a fundamental difference: the difference between a limiter and a maximizer.
Sometimes in mastering suites or in the plugin bundles that comes with a Daw these two concepts are exchangable, like they mean the same thing, but it's not exaclty like that, and today we are going to see why.
We have already covered what a Limiter is: it is a tool that takes a track at mixing volume (usually around -12db) and that sets a ceiling that cannot be surpassed, then with the gain control we can gradually rise the volume until it sounds on par with the commercial tracks (this is if we are using the limiter for Mastering, but we can use it also to tame certain single tracks in a mix).
A Maximizer instead is not only a limiter but it includes also one: it is a tool used to get the most volume out of our recording by applying 2 different processes to our track: a Limiter and a Normalizer.
A Normalizer is a processor that takes the loudest peak of our signal and boosts the whole track until this peak reaches the set ceiling, but the maximizer first applies a Limiter that levels down the loudest parts (according to the a selected threshold) and THEN applies a Normalizer, thus first reducing the dynamic range of the track and then rising the overall volume. The reduced headroom creates more space in which we can pump up our track.
While a Limiter as we have said can be used also on single tracks during the mixing phase, a maximizer is used almost exclusively during mastering, therefore it often features dithering and other tools made to finalize the track: unless you are using it for mastering, you might want to turn these extra features off.
Which one to use? It depends on the type of music: if the music is very uncompressed and presents a lot of headroom, a maximizer can help in reducing the excessive dynamic range, but if the song is already quite compressed (e.g. edm) the dynamic range can be already little, and a maximizer can end up in creating distortion artifacts and messing with the low end, by producing a "pump" effect that is rarely desired.
My final suggestion is to use a Limiter as a general rule, and to switch to a Maximizer only if the mix sounds still weak and too many loud peaks that eats up too much headroom, but be always very careful in leaving enough dynamic range in your song, or it will become ear fatiguing and less enjoyable.
Hello and welcome to this week's article!
Today we are bringing a new topic on our infinite list of article about compression: upward compression!
What is upward compression?
It's a type of compression that works exactly in the opposite way than the regular one: you set a ceiling, and decide a whole wave signal or part of it (with a multiband compressor / expander) in which the volume will be increased until it reaches the ceiling.
To make it easier: we choose a part of the sound, or the entire sound, and make it louder until it reach a certain level.
The signal is increased within a ceiling or within a range (some expander has the control named "range" in which the upper limit is the ceiling), and it just increases the volume below the threshold (rather than lower it as a regulal compressor would), putting it into "the range".
If we are expanding the entire sound until it reaches the ceiling, the result will be the same of a regular compressor: the loudest parts of the sound will be attenuated, the quietest will be raised in volume, that's why this procedure is more suited when it affects only a certain frequency range.
Why do we do this way instead of just raising the volume?
To have more control over our transient;
in facts, if we have a sound that is too much oriented on the high end and we want to bring out the body whilst at the same time mantain controlled the dynamic range, we can expand the low end of that sound (like in the picture, in which only the low mids, the part coloured in blue, is expanded, while the part in purple is normally compressed).
Why don't we just use an equalizer to boost the part we would like to expand?
Same answer: because we want to control the transient and avoid it to have too much dynamic range, with volume spikes that can end up out of control.
The uses of an expander are several: to reshape the sound of a microphoned guitar or bass that sounds too thick or too thin, to give life to certain frequency areas of the room microphone of a drumkit, or to apply corrections in a band recorded with few or one single microphone.
This leads us to the same dilemma we arrived when talking in general about multiband compression:
is it a compressor? Is it an equalizer? Many producers consider it the magic wand to solve every mix and mastering problem, while others prefer using the basic tools (broadband compression, equalizer, volume) following the rule that it's better to use few simple tools and master them rather than experiment too much with more complex tools with the risk of ruining a mix.
Hello everyone and welcome to this week's article!
Today we are going about Mastering, and in particular of RMS levels when using the last plugin of our chain in the stereo bus: the limiter.
As we know, a limiter is a tool that prevents any peak to surpass a certain ceiling, and does this by applying a strong gain reduction that blocks the loudest part of the signal.
We have also already talked about the Loudness war so I won't repeat myself here, what I think is an interesting addition is though the reading of the RMS meter.
In order to introduce this concept we need to explain the difference between Peak Level and Rms Level.
Peak level: this is the loudest peak reached by our track. When mixing it could be -12db, but when mastering we can use a Limiter and stop it at -1db, or -0,1db.
Rms level: root means square level. This level is the AVERAGE loudness of the master, and the difference between the peak level and the rms level is an approximation of the amount of headroom left in our master.
What is a good compromise between a mix that is loud enough and that is not squashed?
A good starting point is to limit not more than 3/4db of peak, but keeping ourself at around -9/10db Rms.
-6.2 Oasis - "Some Might Say": Severe clipping distortion -4.9 Metallica - "The day that never comes" (CD): Massive distortion & clipping -7.7 Feeder - "Pushing The Senses": Heavy clipping distortion -10 Katatonia - "Consternation": Awesome (clean) sound, massive choruses -13.1 Sugar - "Fortune Teller": From 1993 -16.9 Metallica - "The day that never comes" (Guitar hero 3) Needs to be louder !
How do we measure the Rms level? A good free tool to monitor our Rms level is Sonalksis Free G, which offers a master fader and a serie of metering tools (place it after the limiter in the post-fader insert).
And you? What Rms level do you like to master your music? Let us know!
Hello and welcome to this week's article!
Today we will try to understand how compression work, in a much deeper level than how we've seen before, because the idea is to stop using presets as much as we can, and to do every step of sound sculpting - mixing with a purpose, with the complete awareness and control of what we are doing.
As we can see in the image, that wave (let's imagine it's a snare hit), can be divided in four different moments:
A - Attack: the beginning of the hit, that rises up to the peak of the sound
D - Decay: the initial lowering of from the highest peak to the body of the sound
S - Sustain: a part of the sound (which we can consider the "body") in which the volume remains more or less constant for a certain amount of time
R - Release: the final decay of the sound from the sustain part to the silence
Let's now consider also the four most common controls we have in a compressor:
Attack, Release, Ratio and Make Up gain.
Attack: how fast, when a sound surpasses a certain threshold, the compressor kicks in
Release: how fast the gain reduction goes back to zero
Ratio: how strong the gain reduction must be. If 2:1 means that for every 2db that surpass the threshold, the compressor make them sound as loud as 1db.
Make up gain: helps us keeping the overall level of the sound comparable to its unprocessed version, so we can turn it up or down until it sits well again in the mix,
Practical applications: In order to use compression properly we need to ask ourseves the reason why we should add it to our channel insert (and if there is no good reason, we just shouldn't): does our track need more attack? Does out track need less attack and more body? Does our track need some dynamic control? Here are some examples of practical use of a compressor:
- We need more body on our snare: we can dial in our compressor the attack to be fast, and to find a ratio that reduces the attack part of our wave of some db, so that it is comparable to the sustain part: this way the quieter parts of the sound (namely the sustain and the release) have more headroom to be raised in level, and therefore the snare sound will have a quieter initial crack, but a much longer and fatter body. So by squashing the attack, we increase our sustain.
- We need to give more attack to our bass: if we want to raise the attack on our bass (for example to make the sound of the fingers on the strings to pop out more), what we need is to turn down the sustain part of the sound, so that the attack remains high, and then we raise the overall level with the makeup gain. How do we lower the sustain part? by turning the release on our compressor down. So, by squashing the sustain, we increase our attack.
- We need to make a vocal track with a high dynamic range even: if we have a song in which the vocals have some part in which it's just a whisper and other in which the singer yells from the top of his lungs, we have a huge dynamic range, and we need each part to be audible the same.
First off when tracking, it's important to set the gain properly according to the loudness of the part (adjusting the input knob in order to reduce the dynamic range a bit), then in the project preparation phase we could do some volume automation to reduce further the dynamic range before starting compressing, so that the compression will be less aggressive in evening out the levels, and finally we can compress, using a fast attack and adjusting the ratio in order to tame the peaks and let the quieter parts to be audible as all the rest.
As we have seen the approach that needs to be understood in order to use compression properly is a REVERSE APPROACH: do we need to boost the sustain because compared to the attack it is too low? We lower the attack and raise the level of the whole wave, instead of trying to raise the sustain part to match it with the attack part.
Today we are talking about a very interesting tool to compress or limit a sound (under this point of view this article goes under the main category of COMPRESSORS) slighly different from the compressors and limiters interface of today.
In the old times (we are talking of the '60s), gain reduction was not a trend as today: sounds were much less "in your face", recordings were less loud and there was a lot of headroom and dynamic range (on the other hand songs were much less powerful and many nuances were hard to hear because they were really low in volume).
In the early '60s a producer called Teletronix created one of the most famous compressors of the time, the LA-2A Leveling Amplifier, a tube-driven peak limiter which still today is used by some classic studio due to its simplicity and the very particular harmonic enhancement produced.
In 1968, then, a producer called Urei created a peak limiter that still today is considered by many studios a standard, and it is still in production: the 1176Ln Peak Limiter, one of the first compressors / peak limiters (at the time the concept of compressor and limiter was really the same) to be completely solid state, unlike the other models which all were tube-driven.
In the modern times Compressors and Limiters have taken different roads, specializing in different tasks and different phases of the music production, but the origin of both is the Peak Limiter, a tool that originally had just two knobs: Peak Reduction and Gain (or Input and Output).
Peak reduction controls the amount of compression to apply to the signal (as can be seen also from the VU meter), Gain controls the output volume. That's it.
Obviously with the following releases of the 1176, the unit started having more and more controls, to the point that today is very similar to a modern compressor: you can choose the compression ratio among four modes (or press all the buttons together to drive the ratio to the max), you can decide attack and release with two knobs and there are different ways of using the VU Meter.
A typical use of a Peak Limiter is on the DrumOverhead Buss: usually the snare is the loudest drum part picked up by the oh microphones, therefore putting a peak limiter in the OH buss lets us compress the snare sound and raise and make stable the level of the cymbals: this way the cymbals will be more audible and with less dynamic range, and at the same time the snare will pop out less but it will gain in weight and sustain. The only thing to remember is to balance the output of the signal, because since we are adding db of gain IN, we should lower accordingly of the same amount of db the OUTPUT in order to leave the overall level of the track unaltered, just processed.
A peak limiter is also good on single tracks such as Bass or Vocals, adding weight and loudness, if we want to use compression as a tone shaping tool, since this obviously is not the most transparent way of processing our track.
One last great way to use a peak limiter is for parallel compression: this way we can really squeeze a buss with all the drum parts, and then mix this sound back into the unprocessed sound, until we find the desired mix between the two.
Mastering: There are some mastering engineers who prefers the subtlety of a multiband compressor in the mastering buss, in order to be able to tame selectively certain frequences and to make adjustments even in the mastering phase, and some other engineers who prefer to have all the mix equilibrium perfect, and then slam the track with a broadband compressor, leveling equally all frequences (I often prefer this solution, and if there is some frequency that needs to be tamed, I prefer going back to the mixing project and solve the problem from there). With the hardware version obviously the hardware needs to be stereo, otherwise two units are needed, anyway this kind of peak limiter is very good in mastering, because often the harmonic excitement and the warmth that it adds to the song is just what it takes to bring a recording to the next level.
Hello everyone and welcome to this week's article!
Today we're going to talk about a topic that we have kinda overlooked in our analysis of the audio processors, because it's basically the other side of the coin of a Gate, but that actually deserves a deepening: the Expander.
An expander comes usually in the same plugin of a gate, and it is always a derivation of a compressor, that's why most of the controls will be the same: threshold, attack, release, knee and so on.
With the same controls, we can obtain different results:
Compressor: it attenuates the signal that surpasses a certain threshold, with a strenght that we can decide (e.g. is useful to attenuate the occasional peaks of a snare drum).
Gate: it attenuates the signal below a certain threshold, leaving the part of the signal above it untouched (e.g. is useful to eliminate the electric guitar background noise)
Expander: it raises the gain of the part of the sound that surpasses a certain threshold. Basically what it does is the opposite of a compressor (e.g. we have to restore some dynamic range from an overcompressed recording, or we have to make the main element of an audio track to pop out more compared to the others).
So the function of an expander is to make a sound to pop out more and restore some dynamic range, without necessarily attenuating the background sounds.
How do we use it? It's simple: we must find the level which gets surpassed only by the element that we want to emphasize and put the threshold to that level, from there with the "expand" control we raise the gain to the desired level. Obviously we can also refine the setting deciding attack and release times.
Usually there is a Gate bundled on each DAW, but not always there's an Expander.
Here's a small selection of Free Vst Expanders for you to try!
Today we're reviewing a different kind of limiter, the Fabfilter pro L.
When we have extensively talked about limiters, headroom and gain staging and mastering, the plugin always taken as our typical limiter was the Waves L2, an industry standard, but actually in the market there are many more limiters which have the same function, but radically different interfaces and mechanics.
The first thing that captures the attention is the rich wave analyzer that takes most of the ui: to the right there are 2 metering tools: the brightest one is the input level, the darkest one is the output level once we have raised the gain, and the red part is the amount of limiting applied.
In the main window we can see the actual wave, with in red the parts that are being limited in real time, and it is possible to choose the range to show in the window, from the full dynamic range to only the peaks that gets reduced. The gain fader is on the left: just scroll it upwards to add more gain, using all the available headroom, and stop when the moment is right: when only the peaks exceeding the average level are limited, and never more than 3 or 4 db of gain reduction, to avoid losing transients. The output knob in the bottom right lets us set the ceiling, according to the main use we want to do with our song: -0.1db if we will hear it mainly on a cd, -1.0db if the main use will be internet streaming, and so on.
The typical use of this plugin is the same as the Waves L2 explained in the "How to use a Limiter" section of this article, which is linking the gain knob to the output one and raising them accordingly until the gain reduction is the right one for your material (sure there are many good presets too, but especially when limiting, presets are almost useless, since they cannot know exactly the loudness of your song, and they will often distort the material): luckily you can browse through the presets also using a function that will not affect gain and output level, just to see how the different settings sounds.
If you want to manually tweak the settings, as in the other Fabfilter products there is an "Advanced Tab" which lets us manually tweak attack, release, lookahead and other parameters as the "limiting algorithm": transparent, punchy (good for single tracks), dynamic (good for preserving transients), allround.
All in all, another very good and complete product from the Fabfilter team, which also integrates a complete monitoring system that will make our mastering chain thinner.
Hello and welcome to this week's article!
Today we are going to talk about de-essers!
A deesser is basically a compressor which is set to operate only on a very narrow range of frequences that corresponds to the sibilants of the human voice, which can be captured with high peaks of volume by the microphones.
Usually these sibilant "S" and "T" sounds are located in an area between the 9khz and the 15khz, and often the deessers have presets for male or female vocals, but the best ones sometimes features also a "learn" mode in which they automatically detects the right area to attenuate: the more the area is narrow, the more the deesser will be effective without processing unneeded areas of the mix.
There are also other, more creative ways to use a deesser: someone uses them on a cymbals track or on a distorted guitar track to tame the harsh frequences that sometimes can get a bit too prominent, giving a more smooth result, similar to the one you'd get from a Low-pass filter, but less invasive.
There are many Deessers on the market, and almost every DAW has one bundled, but if you want to try some free Vst, here are best ones the web has to offer:
Hello everyone and welcome to this week's article!
Today we're going to talk about another Fabfilter product, the Pro C Compressor.
By now, after all the articles I've done, we all should know what a compressor is and how does it work: basically it's an automatic volume control that lowers the lowest peaks according to how we set it, and raises the quieter parts.
I have particularly liked this specific compressor because of its flexibility, which is similar to the Pro Q 2's: the interface is clean, easy, and at the same time there are tools to visually monitor what's going on that usually are not featured on other processors of this kind: I'm talking about the dual graph:
- the left one is pretty common on many compressors, and tells us when and how strong the compressor kicks in, and it is possible to choose between a soft and a hard knee.
- the right one is less common and it shows us 3 things: the original waveform, the amount of gain reduction we're applying, and the new waveform. This helps us a lot checking out if we're crushing the dynamics of our track too much, or on the contrary if the settings we're applying are uneffective.
The last part of the graph is a metering tool that tells us the level of our track and the gain reduction applied.
Beside the common controls that we can find on any other compressor (attack, release, ratio..), the Fabfilter Pro C features also some less common function:
- Automatic Release control, that changes adaptively according the part, and that is particularly useful if used on complex tracks
- Automatic Make up Gain to compensate the usual loss in volume due to the compression
- 3 Modes: Clean, Classic and Opto, which reacts differently, since they're modeled on different types of compressors.
- the A/B comparison function: we can set the compressor with two different settings and a/b compare them to choose the best one.
- the Parallel Compression function: we can experiment using some more extreme setting on our track, and then use the Dry Mix control to decide the amount of dry signal to blend with the processed one. This way we can add some body to the track leaving part of the transient to pass through uncompressed. This is particularly good with a snare drum, to keep the snap (the transient) intact, or with a whole drum buss.
- a Sidechain Function in which we can decide wether to use separated left and right gain control, or to use it Mid/side.
- a Midi Learn function to automate the settings: we need to push the Learn button, then move the knobs we want in real time while the song is playing, and the Pro C will record our movements and change the settings real time the same way we did, automatically.
There are even more functions to talk about (for example this plugin can be used as an effective Deesser too), and many presets to choose from, but it's already enough to say safely that this is the most complete Compressor ever made, and one of the most transparent sounding ones, too.
Hello and welcome to this week's article!
This time we're going to expand our article about Mixing with Multiband Compression talking about the main purpose of this tool: Mastering.
When the track is ready for the mastering phase, we all know that it should pass though the last processing steps before reaching a volume comparable to the commercial tracks, adding if needed the last "icings on the cake", but it is actually very important to take care of the balancing and the other aesthetic aspects of the song in the mixing phase, resolving the issues before arriving to this last step:
the track should already sound at its best, leaving to the mastering phase only the final push.
Looking things under this point of view it appears crucial to work in the most transparent and essential way, because the risk of compromising the complex castle of cards that is a perfectly balanced mix is very high, and this is the reason why some mastering engineer prefers putting in the mastering chain a Multiband Compressor instead of a Broadband One: to assign a particular setting to each part of the spectrum, instead of using one processor that affects with the same intensity the whole mix.
Let's make an example of how to use a multiband compressor on our master track, and for this example we're going to use the Waves Linear Phase MB, as in the picture above, but basically the same mechanics applies to any other multiband comp.
Let's open the Waves Linear Mb and play the whole song, adjusting the input looking at the top right corner meter in the UI, so that it doesn't clip (we can also click on the "Trim" button to lower the gain right below the clip level).
When the whole song will be played, we'll check out the number on the right of the "Solo" and "Bypass" buttons on each band. These numbers are the peaks on each band, so we will need to dial these same numbers in the "threshold" section of that same band (for example -10, -5,5, -12 and so on).
Once we have done this task, we will notice that the sound should be more "stable", since only the infrequent highest peaks that reach that level will be tamed, and only in their most relevant frequency area. This operation will give us some extra headroom to raise the overall level, and ultimately, will let us obtain a more powerful and transparent master, if we don't love the slighly more invasive feel of a broadband compressor.
Now it's time for the final adjustments: we can add or lower some db of gain on the single bands if we need (this will combine the compressor function of this tool with the tasks of an equalizer), and we can fiddle around with the Attack and Release functions, remembering that usually you have set them properly if you see the compressor kicking in and out in time with the song.
It is also possible to turn on or off the "Makeup" function, which raises the gain according to the gain reduction produced, and to set the "Adaptive" level, which helps avoiding one frequency area to mask another one through the song.
Hello and welcome to this week's article!
This time I would like to share with you my thoughts about compression, and its dual role in the mixing phase.
So far in our many mixing tutorials we have seen compression for its main purpose: to lower the volume of the loudest parts of a sound, thus creating room to rise the quieter parts; this is the concept of Compression used as a gain reduction tool, the way it was created in the first place, and it is explained in the compression tutorials.
The other role of this tool is to shape the tone, that's why they say that equalizaton and compression are the two main tone shaping tools, because just with them we can create modify the source sound in the 90% of what we have in mind, the rest that cannot be done with them is to be considered just the icing on the cake.
With "tone shaping tool" I mean that we can use the compressor not only to intervene with the gain but also to modify the structure of a sound and make it better, in a way similar to the equalization.
This concept applies especially with drums and vocals: there are times in which we would like to give fatness to a sound and to make it more similar to what the radiophonic rock songs sounds like, and we use the equalizer trying to pump up the lower mids, with the results that our tone loses clarity and becomes boomy, while we could just use compression in an intelligent way, and we would get much closer to our target even without touching the eq. Drums and vocals are instruments with a very high dynamic range, and if we don't compress the sound wisely we will end up hearing just the loudest frequences and lose all the rest of the content, which is what gives "the body" to a sound: think about the snare sound, its original tone is usually snappy and ringy, but we need it to make it sit in the mix in a stable way, and at the same time to make it sound fuller, and with some tail similar to a gunshot: if we start compressing it, beside the evident gain reduction that must be compensated, we will notice that the sound will gradually lose its snap, and will become more and more explosive until, if we keep turning the knob, the sound will become an almost inaudible "oomph".
Our target would ideally be somewhere in between, with the snare stable (which means that it doesn't disappear nor covers the other instruments), and with the right amount of snap and explosiveness.
Radiophonic rock producers tends to compressdrums very much to leave the sound full and to make it always audible, even if the radio signal is low, and to obtain this they often kill most of the dynamics, but it's a trade that's worthy, speaking about music that will often be heard in the car stereo, with sub-par signal-to noise conditions. Compression can be used as a tone shaping tool also for bass, giving it attack, roar in the lower mids and clarity, and to vocals, fattening the sound without intervening with the additive eq, that most of the times can damage the sound, somehow.
In order to use Compression as a tone shaping tool, the producers have also created Multiband Compression, which is a step in between a comp and an eq, and it's aimed mainly to compress some areas avoiding damaging other parts of the spectrum, and to modify only the tone of some parts of the mix/master.
I hope this helped you understanding some of the collateral functions of this powerful tool!
Hello and welcome to this week's article!
This article links our in-depth Mastering article, which covers pretty much everything that can be done in the mastering phase, whith our "Mixing with stock plugins" article, which shows us how to mix a song only using the basic stock plugins featured in our daw.
The idea is to try to separate what's essential when mixing and mastering from what's just "the icing on the cake", and to focus on the essential things with just the tools provided in our Daw.
The second objective is to free ourself as much as we can from the slavery of the presets: we must understand how does a tool works and how to adapt it to our song; the presets that we can find on the plugins are always a caricature, made to let us understand the direction of the processing, but they're always exaggerated.
The sooner we'll stop using presets without knowing what the hell is going on, the better our songs will sound, trust me.
Speaking of the essential mastering chain, once we have a good mix we just need to give our tracks the final boost in volume and the final touches to make them sound in line with the commercial products.
Let's start with a compressor. This is needed to reduce the dynamic range a bit, letting us push more with the limiter. We need to check out the average level of our song (for example -10db) and set the threshold of our compressor a bit higher (for example -8db), so that the compressor only lowers the big peaks. The Ratio should be somewhere between 2:1 and 8:1, we need to limit the peaks but not taking away the life from our songs so find the right compromise. The attack and release should be set to a speed that lets the compressor to kick in and go back to inactivity, we don't want to compress everything all the time (ideally, the meter should move in time with the song).
Now we need to use an Equalizer. In theory, in the mixing phase we should have already done all the equalization we need, and if there are problems it would be better to re open the mix and to fix it from there, because now we're affecting the whole mix, so we must be very careful.
A good thing to start with the mastering eq is to take away everything from 45/50hz with a high pass filter, then we can check out the most problematic areas in the mastering phase, which are usually the lower frequences: sometimes our master needs a little boost in that area, sometimes a little cut, sometimes nothing (and we often can notice it only using a reference track), but probably the best way to eq a track during mastering is using a Mid/Side processor, in the way described HERE.
Now that we have tamed the most problematic frequences we can try to add a little sparkle to our sound, with some harmonic exciter or saturation tool. The ideal would be to have a multiband processor, since in my opinion we should excite a bit more the higher frequences and much less the lower ones (or just bypass them), otherwise the delicate equilibrium we've tried not to ruin until now will be damaged. These tools are usually needed to add some nice ringing to the snare and cymbals, and to add some bite to vocals and guitars, but if we see that we cannot obtain a sensible enhancement, we can just skip this whole step and pass to the limiters.
The limiter is the single most important part of the mastering chain, as we have already seen in many other articles of this blog. What we need to do is to see how much headroom we have and to raise the final volume of our song to a point that it is competitive with the volume of the commercial songs, but at the same time it must not sound excessively squashed and distorted. Our job here is to find the right compromise between power and clarity, without cutting too much the transients.
We should set the ceiling somewhere between -0.1db (if our final track will be played mostly from a cd) and -1.0db (if we are planning to distribute it mainly through internet streaming) and to lower our threshold until we see some gain reduction: we should let the limiter kick in only with the highest peaks, and never let it limit more than 3 or 4dbs, otherwise we will damage our song.
In order to know if the overall level of the song is in line with the industry standards and if there is enough headroom before exporting, we could use some metering tool. Those tools just tell us visually if the song volume is right, or if we could raise it or lower it, and some tool like the TT Metering tool also tell us if the song is too compressed, so that we can limit it less and make it less ear fatiguing.
Once all the meters says that the levels are right and there are no distortions, we can consider our mastering done!
Hope this was helpful! By the way, the song on the video is a song taken by Wisteria's full lenght "8-Bit Nightmare", played and mixed by me. Buy one copy to support us by clicking on the banner on the right side of this blog!
Hello and welcome to this week's article! We have finally approached the end of the year and to the end of this long tutorial on the many ways to mix a rock / metal drumset!
The snare is the focus of the album, along with vocals, and the equilibrium among vocals, snare and kick it's the axis of the mix, the one that sets the mood of the song (higher kick and snare and lower vocals for death metal, higher vocals and lower drums for soft rock): long story short, we need to get the volume and the tone RIGHT, otherwise the whole album will sound cheesy.
If we have just the snare top track, we can start by using a gate, to take out at least part of the cymbal bleed (not too strong, or we'll take away also some of the good harmonics of the snare), then a hi pass filter up to around 70hz and finally a low pass one down to 12khz, to cut away some of the left cymbal bleed.
Now we need to take away the low-mid "mud", which lies among 200 and 500hz: just find the right frequences and take them slighly down, without taking away too much body from the sound (you can use a frequency analyzer to locate exactly where most of the energy lies).
Now if the snare needs some low-end boost we can lift with a medium-width boost the area around 125hz, but beware because we risk to end up by adding again the "mud" we have just taken out.
The last and trickiest thing to do to enhance the tone with an equalizer it's to boost the highs: if we feel that the snare sounds too dark we can try boosting somewhere between 2'000 and 6'000hz: this is the most delicate moment of all, because now we are defining the colour of the tone that will hit the ear of the listener first, so we must not overdo and be aware that the higher the frequences that we're boosting are, the louder will be the cymbal bleed.
Once we have found the right tone for our snare top track we can compress it at a ratio up to 8:1 (we must choose the ratio according to the dynamic range of the track: if we have many low energy snare parts like press rolls, flams etc, we need more compression or they will disappear in the mix, while if there are not and the strenght of the hits is consistent, we can lower the ratio).
The attack should be around 10ms, and as always we should set the release at a level that the compression can go back to zero or almost, between a hit and the other.
If we feel that by compressing we have lost part of the transient we can use a Transient Shaper to bring back some of the snap of the snare, usually avoiding cymbal bleed, but don't overdo because too much transient enhancement can make the snare too much "in your face" and unpleasant: we must always look for that euphonic "sweet spot" :)
If we need to transform the snare sound too much to get a decent tone, then we have a problem: an excessively processed snare it's a snare that sounds fake, therefore it would be a good idea to record it again. If we can't record again and we need a way to add some punch on the snare we can mix one or more samples with the original sound, and if we're good, we will find a sample that compensates the lack of frequences of our original sound.
Another way is to Double the snare track: we can process the first one like explained above, and use the second one just to add the "snap". We can gate the sound until we have only the snare snap, then we compress it very heavily, use some frequency exciter and eq to make the "snap" very strong, similar to a fingersnap, and then we can mix this sound with the regular snare to get the same effect we would obtain by using a sample, with the difference that this way the sound comes all from our drumset.
Finally, if we have recorded the snare with 2 microphones, top and bottom, we're gonna have a second track recorded pointing to the snare wires in order to give some sizzle to complement the top snare tone. This track can be also very useful to add some body to the top snare tone: instead of boosting around 125hz on the top track, we can boost the same area in the snare bottom track and see if it sounds better. About the hi pass filter, we can take out a bit more, even up to 90hz if we want, the important thing is to take a look at the phase: we must try inverting the phase to see the version that produces less frequency canceling, and use that one.
On the compression side, if we use more than one snare track (top + sample, top + bottom, top + top copy), my suggestion is to route them into a group track and compress that one, with the same settings suggested for the snare top.
If we need it, we can also set a clipper at the end of the snare chain: this is sometimes useful because, during the mastering phase, the limiting traditionally takes out part of the snare transient, but if we use a clipper (with very soft settings) we can recover part of that lost transient (it's not easy to explain how a clipped sound helps recovering its transient while it should actually work exactly at the opposite, but it works: the software recreates part of the transient).
It's interesting to notice that unlike for other instruments (e.g. hi gain rhythm guitar), in which the compressor should be as transparent as possible and the moment you notice it, it means that is damaging your sound, compression for the snare is a real tone shaping tool, and if used without exceeding it can bring up the wires of the snare bottom, and enhance the sound with an explosivity that cannot be obtained otherwise.
About the Reverb: if our drumset sounds a bit dry (or we like a big '80s style snare) we should create an fx track with the best reverb that we have and set it in order to make it sound similar to a studio room.
Then we must equalize it with a high pass filter that takes out everything up to around 200hz, and send it to our snare track (or group), to our tom group and to our cymbalgroup track, from there we can decide the amount of effect sent to each track according to our taste.
I hope that this long drum mixing tutorial was helpful, and by the way the song is mixed and mastered by me for my band, Strider. Contact us if you want a copy of our latest Ep "The Black Lotus", also to support my work with this blog! Happy New Year from Guitar Nerding Blog! CLICK HERE TO READ PART 1/4
