domenica 26 febbraio 2012
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, but if our music will be listened mostly on mp3 or Youtube, it's better to set it at -1.0db, in order to avoid unwanted distortions.
- 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. To learn More about the Dynamic Range, and how to measure it, check out our article about the Loudness War.
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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domenica 19 febbraio 2012
HOW TO USE SIDECHAIN COMPRESSION (free Vst Plugins included)
Hello and welcome to this week's article! Today we're going to talk about an advanced compression technique called "Sidechain Compression", which is widely used in productions that span from extreme metal music to dubstep. To take a look to the Basics of Compression, CLICK HERE.
Sidechain Compression is a technique that consinsts into putting a compressor on a sound (usually the kick), that when is triggered, it applies the compression on another sound (usually the bass, or the synth on dance music), instead of affecting the kick itself;
the aim is to obtain a "pumping" drum sound, almost as the bass is part of the release of the kick itself (therefore largely increasing the groove), and avoids at the same time certain frequencies of the bass to cover the same frequencies of the drum kick, leaving it as much clear and audible as possible.
Let's focus on its most common use, for rock-heavy metal music, and once you've learned how it works, you can try it on more creative ways.
First off, grab a vst compressor that supports sidechaining, Here is a List of Free Sidechain Compressors, and among them I obviously suggest the ReaComp; many DAWs, anyways, already have an "in bundle" compressor that supports sidechaining.
Now load the compressor on your Bass track and activate the "sidechain" control (for example, in Nuendo, it's an orange button on the top left of the effect window, if you lay the cursor on it without clicking it will say "Activate Side-Chain"). Once this control is active, open the kick track and find the "Audio Sends" menu. By clicking on one of the empty slots, it will make you choose the buss where to send your Input, and choose "Sidechains - Reacomp", or whatever is the compressor you've loaded on your bass track. After you chose, set the "Send Level" to 0.00 db.
Now, by playing the two tracks together, you will notice how the kick will cut through the bass sound much clearer, and the more you will lower the "Threshold" control, the more the bass volume will be lowered when the compressor activates (the ideal would be that when the kick hits, the bass is lowered of around -7db).
The attack time should be set pretty fast, according to the music genre, and so the release time, that may vary orientatively from 250ms to 500.
sabato 11 febbraio 2012
HOW TO USE MULTIBAND COMPRESSION WHEN MIXING (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 (CLICK HERE FOR AN ARTICLE ABOUT USING MULTIBAND COMPRESSION IN MASTERING). 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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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 (CLICK HERE FOR AN ARTICLE ABOUT USING MULTIBAND COMPRESSION IN MASTERING). 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.
domenica 5 febbraio 2012
HOW TO USE PARALLEL COMPRESSION (free Vst Plugins included)
Hello and welcome to the advanced part of my Compression Tutorial (for the Basics of Compression, click HERE).
Today we're going to talk about Parallel Compression, which is a particolar way to apply this effect, in order to achieve certain results which are heavily used on professional studios.
PARALLEL COMPRESSION:
This technique is commonly used for drums or vocals, or even for the whole mix. It consists in creating on your D.A.W. project an Effects Track (or FX Track, or Fx Buss, there are different names according to the software used) or a Group Track, loading on it a Bus Compressor and/or an equalizer or any other effect you may need, and blend this track with the other already-processed tracks that you have.
A drum group is perhaps one of the best examples of when we would use Buss Compression, and since it’s very likely that you would insert individual compressors on some of your separate drum tracks, we can call it Parallel Compression (since e.g. the kick, which you may have already compressed on its own, will be routed along with the other drum parts on the Fx Buss which contains the compression, and the two compressions will be summed).
Let's try this experiment: Record a drum kit, or use a virtual drumset, and apply some equalization and compression on the single parts you need, then create a Stereo Group Channel, load a compressor on it (for example the Audio Damage Rough Rider), and set it very aggressive (threshold -40dB or so, ratio set to infinity). The idea is to blend this heavily compressed sound with the original, softly processed or even unprocessed one, to retain some of the track's original transients and musical dynamics.
I prefer a long release time (maybe around half a second, more or less), but the real secret to making this work is how you set the attack time, we can start setting it somewhere in between 2-10ms, but it's really all a matter of experimentation and different tastes. Once you've found a sound you like (you can also add an equalizer and try to boost 6db around 100Hz and 10Khz), open your snare, kick tracks and all the others you wish to effect, go to the Fx Send menu of the single tracks, and load the Stereo Group Channel you've created with the Compressor.
From Here, you can blend the effects track with the single instruments tracks, choosing the amount of effect that will affect the single drum tracks.
This compression technique is also known as the New York Compression Trick.
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sabato 28 gennaio 2012
HOW TO USE EQUALIZATION (free Vst Plugins included) PART 2/2
Finding the right frequency to adjust is, of course, the most important thing. With time you will get familiar with some of the most common frequencies, but what if you’re dealing with a new sound, or just don’t have the experience to know where to start? Here is an easy way to find the right frequencies to deal with.
What you need is an EQ that allows you to control the target frequency. Boost one band all the way. If the band has a “Q” control make it quite high (Q stands for “quality factor” and it controls how wide the area affected by the boost or cut will be).
Then, play the sound and slowly sweep the frequency back and forth until you find the point where the tone you are looking to focus on is loudest. Make a note of the frequency and put the EQ back to zero. You now know the frequency where your target tone occurs and can cut or boost appropriately.
It’s a good rule that when cutting it’s best to use a narrow (high) Q, while it is better to have a wide (low) Q when boosting. This will help keep your EQ not excessively invasive.
The main reasons why it is better to cut than to boost is that excessive EQ boosting in a mix usually results in muddiness and loss of clarity, so as we already said, use eq with parsimony.
The low-mid range is where all the fullness and body lies for most of the instruments. For this reason it can be tempting to give those instruments plenty of low-mids. The problem is that all those low-mids fight for room in the mix and if you aren’t careful you’ll be left with an engulfed, unintelligible mess.
Think of your mix as a physical space: the more instruments you put in your mix, the harder it will be to fit everything in.
Therefore, more instruments in a mix, the more important it becomes to keep an eye on the areas where their tonal ranges tends to overlap. Each instrument needs its own place to sit in the mix, so any time there is a common range you need to choose which instrument takes the forefront in that frequency range.
For instruments that have the same basic range, such as bass and kick drum or two guitars, you can use the EQ to interlock them making them both distinct, unless you want a particular effect, as the "rhythm guitar wall of sound".
Talking about frequencies, and how to use them for single instruments, first off let's say that on the website RECORDINGEQ.COM there is a very interesting page with many standard frequencies, and the explanation of how they affect the mix, I think it's a great starting point as a reference, so check it out.
If we want to simplify, though, here are some eq references to begin with:
9KHz-15KHz:
Adding will give sparkle, shimmer, bring out details.
Cutting will smooth out harshness, and darken the mix.
Common processing: Very little compression, add/reduce gain for
timbral shaping.
6KHz-15kHz:
Air and presence.
Common processing: Slight gain boost.
4KHz-9KHz:
Brightness, presence, definition, sibilance, high frequency distortion.
Common processing: Compression to reduce sibilance/HF distortion. Add
(gain) brightness or liveliness to a mix.
5Khz-7KHz:
De-essing. Narrow band compression.
800Hz-4KHz:
Edge, clarity, harshness, defines timbre.
Common processing: gain reduction to reduce harshness.
200Hz-1.5KHz:
Punch, fatness, impact.
Common processing: Compression and gain boost.
150Hz-400Hz:
Boxiness.
Common processing: Reduce gain to remove 'mud'.
200Hz-below:
Bottom.
Common processing: Compression to tighten a boomy bass sound.
(source: http://www.dogsonacid.com/showthread.php?threadid=25008)
If we want to simplify, though, here are some eq references to begin with:
9KHz-15KHz:
Adding will give sparkle, shimmer, bring out details.
Cutting will smooth out harshness, and darken the mix.
Common processing: Very little compression, add/reduce gain for
timbral shaping.
6KHz-15kHz:
Air and presence.
Common processing: Slight gain boost.
4KHz-9KHz:
Brightness, presence, definition, sibilance, high frequency distortion.
Common processing: Compression to reduce sibilance/HF distortion. Add
(gain) brightness or liveliness to a mix.
5Khz-7KHz:
De-essing. Narrow band compression.
800Hz-4KHz:
Edge, clarity, harshness, defines timbre.
Common processing: gain reduction to reduce harshness.
200Hz-1.5KHz:
Punch, fatness, impact.
Common processing: Compression and gain boost.
150Hz-400Hz:
Boxiness.
Common processing: Reduce gain to remove 'mud'.
200Hz-below:
Bottom.
Common processing: Compression to tighten a boomy bass sound.
(source: http://www.dogsonacid.com/showthread.php?threadid=25008)
Focusing on ROCK/METAL GUITARS, my first suggestion is to try to get the right eq before recording, or directly on your guitar simulator, in order to modify it with vst equalizers in the most subtle and less invasive way possible.
First thing, let's remove everything you’re sure you won’t need, for example for someone, frequencies around 100hz are useless for a metal guitar, since it's a boomy area and taking it out will increase clarity. Next are the hiss, or fuzz frequencies which usually lie around 5050k, 8k, and 9650k. Dipping these frequencies by a few DB will take out the hiss, especially created by digital amp simulators, but can take some tone with them at times. Remember; it’s all experimentation!Someone prefers even to cut everything above 7khz (or more) to get a more mid-focused sound. To me it looks a little too much, better to low pass the sound starting by a higher frequency, like 12khz, and to high pass above 80hz.
Now we may add some EQ, lowering a little bit around 250hz and 650hz for a deeper scoop of middiness (if needed), and some 8k dip for clarity, to get rid of some of the boxiness.
About the frequencies to boost, it obviously depends on the type of sound, and more than often a good guitar sound doesn't need any boosting at all, but sometimes I found that adding a couple of dbs around 200hz and 3235hz may help to achieve a more aggressive sound.
Here are some of the best Free Vst Equalizers around:
Sonimus SonEq
http://sonimus.com/site/page/downloads/
Windows: VST
Mac: VST & AU
SPL Free Ranger
http://plugin-alliance.com/en/plugins/detail/spl_free_ranger.html
Windows: VST, RTAS, AAX
Mac: VST, AU, RTAS, AAX
Elysia Niveau Filter
http://plugin-alliance.com/en/plugins/detail/elysia_niveau_filter.html
Windows: VST, RTAS, AAX
Mac: VST, AU, RTAS, AAX
BX Cleansweep (Hi&Lo Pass Filter)
http://plugin-alliance.com/en/plugins/detail/bx_cleansweep_v2.html
Windows: VST, RTAS, AAX
Mac: VST, AU, RTAS, AAX
Variety of Sound Boot EQ mkII
http://varietyofsound.wordpress.com/vst-effects/
Windows: VST
CLICK HERE FOR THE PART 1/2 OF THIS TUTORIAL
CLICK HERE FOR A GUIDE ABOUT HOW TO RECOGNIZE THE FREQUENCIES!
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About the frequencies to boost, it obviously depends on the type of sound, and more than often a good guitar sound doesn't need any boosting at all, but sometimes I found that adding a couple of dbs around 200hz and 3235hz may help to achieve a more aggressive sound.
Here are some of the best Free Vst Equalizers around:
Sonimus SonEq
http://sonimus.com/site/page/downloads/
Windows: VST
Mac: VST & AU
SPL Free Ranger
http://plugin-alliance.com/en/plugins/detail/spl_free_ranger.html
Windows: VST, RTAS, AAX
Mac: VST, AU, RTAS, AAX
Elysia Niveau Filter
http://plugin-alliance.com/en/plugins/detail/elysia_niveau_filter.html
Windows: VST, RTAS, AAX
Mac: VST, AU, RTAS, AAX
BX Cleansweep (Hi&Lo Pass Filter)
http://plugin-alliance.com/en/plugins/detail/bx_cleansweep_v2.html
Windows: VST, RTAS, AAX
Mac: VST, AU, RTAS, AAX
Variety of Sound Boot EQ mkII
http://varietyofsound.wordpress.com/vst-effects/
Windows: VST
CLICK HERE FOR A GUIDE ABOUT HOW TO RECOGNIZE THE FREQUENCIES!
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sabato 21 gennaio 2012
HOW TO USE EQUALIZATION (free Vst Plugins included) PART 1/2
Hello! Today we're going to talk about equalization.
The job of this effect is to correct the timbric shape of the sound, operating on its frequencies. Usually with the term "equalizer" we can refer to any tone shaping control, and on the guitar amplifiers we usually have it under the form of pots, often divided in lows, mids and high frequencies.
First we can divide equalizers in two groups: active and passive.
The passive equalizers are nothing but filters that limit the passage of the selected frequencies, so they can not amplify a certain frequency but only cut. Most of common guitar amplifier equalizers are passive.
Active equalizers, instead, are the ones that attenuate or amplify a selected frequency, so they can be not only used to cut but also to boost certain frequencies (those are more rare on guitar amps, but some amp, like the Peavey Triple XXX, has them).
Then we can divide equalizers under a functional viewpoint in three groups: Analytic, Graphic and Parametric.
Analytic equalizers support a regular boost-cut control, and an extra control that gives the ability to move upper or lower on the audio spectrum the central frequency of the regular control. Usually we can find this on the mixing boards, and an example of it may be the free REAEQ vst.
Graphic equalizers were formerly created for the recording studio, but today they can be found on many guitar amps (for example Mesa Boogie Mark IV), or as a stompbox or rack effect.
A good example of graphic equalizer is the free KJAERHUS Classic Eq Vst.
Parametric equalizers are usually different: their interface is made of pots instead of a graphic representation of the spectrum, therefore forcing you to rely even more to your ears, and they supports a control to shape the width of the frequency modifications, called "Q". A good example of this approach to the matter is the free PUSHTEC EQ Vst.
When to use Equalization: EQ should be the last resort. This said, you should try to get your tone as perfect as possible right from the beginning: if you’re not completely happy with the tone you’re getting without an EQ, then keep trying different recording strategies before deciding to use it, as it will unavoidably affect the realism of the sound.
When recording with microphones, their placement can be one of the biggest factors here: don’t be afraid to spend the time trying as many different placements as necessary to get the right tone.
Keep in mind that small changes in placement can make a big difference, as for example pointing the microphone right towards the center of the cone or to some other point halfway between the center and the border, or keeping it perpendicular to the cone instead of setting it angled.
If you have great tone from the start, then EQing during the mixing process will be little more than gently pushing the sound into its place on the mix.
Beware of Frequency Masking: Frequency masking is where the frequencies of two (or more) instruments are fighting for the same place in the spectrum, and it is one of the most common problems encountered when mixing.
In facts, a sound may have (in addition to its root frequencies) other Harmonics that contribute to its overall timbre. If two sounds share similar frequencies you could easily find yourself in the position where some of these Harmonics are being masked in the mix; meaning that the instruments sound different than they do in isolation, or just disappear, and this problem emerges with more clarity while listening to the mix in mono.
There are three ways to deal with eq masking problems:
1 - Try to Pan the two sounds on different places of the soundstage, and see if the problem is solved.
2 - Create a priority list of your tracks, and start carving some db from the troublesome are of the "less important tracks", until you can hear with more clarity the primary tracks (e.g. Vocals).
3 - When the aforementioned operations are impossible, the last chance is to modify the arrangement of the song, taking out one of the fighting tracks from the part where the mix becomes too crowded to be heard clearly.
CLICK HERE FOR THE PART 2/2 OF THIS TUTORIAL
CLICK HERE FOR A GUIDE ABOUT HOW TO RECOGNIZE THE FREQUENCIES!
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sabato 14 gennaio 2012
HOW TO USE IMPULSES (free VST plugins included)
Hello and welcome to another post of this blog! Today we're going to talk about Convolution Impulses.
Convolution impulses are digital simulations of the reverberation of a physical or virtual space.
Why do we need them? Because today those are the best replacement for guitar cabinet simulation. In facts, you may have noticed that the main issue of Guitar Amp Simulators (both hardware, like Pods, or software, like Amplitube) is the cabinet simulation and microphone response: they sound harsh, fizzy and plastic, because although they are impulse-based, their response is not accurate enough.
The web nowadays is full of impulses free to download and to experiment, and many of them are really accurate and realistic, that is the reason why many studios use them instead of microphoning a real cabinet. They are great for Reamping too, and can be used with any kind of preamp (a preamp vst, a digital multieffect or a real tube head, through its preamp out), taking the place of power amp, cabinet and microphone.
Just connect the preamp/pedal output into your soundcard, load one of the programs made to play them (for example on this guide we're gonna take a look to VOXENGO BOOGEX), load an impulse and play: if you hear a normal tone, with no harshness/fizzyness, you're done. WARNING: Do not connect the power amp output of your head to your audio interface! Use the preamp output or the Fx-Send. If you have a tube power amp, you have to connect it to the cabinet anyway in order to prevent tube damages!
Here's how to load an Impulse on Voxengo Boogex:
First off turn off the cabinet simulation of your software or hardware (for example, with Amplitube 2: go into the cab section and activate the BYPASS switch you see in the upper right corner, or with the Pod Hd just take the cab icon away from the chain of your preset, setting it to No Cab), if you have bypassed the cab correctly, when you play a distorted preset you'll hear an awful sound, fizzy as hell, no bass, harsh... You're hearing how an amp sounds like without a cab.
Now load Voxengo Boogex (I chose to explain this among the others because it's free, not too heavy for the ram and with zero latency, but there are many others around for free, as KeFir),
Boogex is an amp-sim, if you want to use it ONLY for cab simulation, you have to set all the parameters FLAT, or it will change your eq/sound.
So, open it and set, from the left: Lows Mids and Highs to 0db, Tone to 0%, Drive to 0db, Dynamics to 0%, Phase to 0%, The "Out" control it's the volume control, Dry Pre Cab and Dry Post Cab to "-inf" (knob all the way down).
The "Speaker Cabinet Impulse Response" button MUST be ON, obviously, then click on "file" and select your impulse (a .wav file).
Another interesting thing about Boogex: this software features a built in Low Pass Filter and High Pass Filter. Since they can't be totally bypassed, the best thing to do is to open them to the max, by clicking on the green point in the right area of the graphic section of the interface and move it all the way to the right, then click on the white point in the left area and move it all the way to the left. This will deactivate the filters.
Boogex is an amp-sim, if you want to use it ONLY for cab simulation, you have to set all the parameters FLAT, or it will change your eq/sound.
So, open it and set, from the left: Lows Mids and Highs to 0db, Tone to 0%, Drive to 0db, Dynamics to 0%, Phase to 0%, The "Out" control it's the volume control, Dry Pre Cab and Dry Post Cab to "-inf" (knob all the way down).
The "Speaker Cabinet Impulse Response" button MUST be ON, obviously, then click on "file" and select your impulse (a .wav file).
Another interesting thing about Boogex: this software features a built in Low Pass Filter and High Pass Filter. Since they can't be totally bypassed, the best thing to do is to open them to the max, by clicking on the green point in the right area of the graphic section of the interface and move it all the way to the right, then click on the white point in the left area and move it all the way to the left. This will deactivate the filters.
Here are some great free impulses to load:
If you have or know some other interesting impulse to suggest to our readers feel free to tell us!
Additional Awesomeness: some speaker simulator plugin (such as Poulin's LeCab2) gives you the opportunity to load more than one impulse and blend them together, as if the speaker is microphoned with two or more mics. This opens up a very interesting world of variables, if carefully managed, but can also harm your final result, so use it carefully :)
Additional Awesomeness: some speaker simulator plugin (such as Poulin's LeCab2) gives you the opportunity to load more than one impulse and blend them together, as if the speaker is microphoned with two or more mics. This opens up a very interesting world of variables, if carefully managed, but can also harm your final result, so use it carefully :)
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