Introduction
Note: this entry was already published in my old Spanish version of this blog, around March 2011. I am just revisiting it here.
From all the pedal effects I've ever tried, compressors were always the most arcane and mysterious. I wasn't never able to understand what a compressor is and, for what or when to use it.
Only when I've started my own home studio and had to deal with tracks and their Peak and RMS levels is when I've started to understand what a compressor does.
Some time ago, I bought a T.Rex Comp Nova, just to have a "complete" pedalboard that "should" include a compressor effect. Doesn't it comes with any multi-effect pedalboard?, it should be important, then!. You know...
Now, with clear ideas, I've decided to give an oportunity to this kind of effect and, I wanted to compare that pedal (bought because of its transparency and dummy-proof controls) against one of the mythic compressors: the MXR Dyna Comp.
And that's all about this entry: how a compressor works, what's its use and, a comparison between a very transparent one (Comp Nova) and a coloring one (Dyna Comp).
What the heck a compressor is useful for?
Let see a very typical situation.
Someday, we see that one of our friends loads a compressor on his pedalboard and, that he/her even uses it so, with lots of curiosity we ask our friend to let us to check that effect.
Usually, he doesn't known how to explain what it does further than "it compresses the sound" or "gives some sustain" or we don't understand a single thing about what he/she is talking about.
Anyway, we switch on the pedal, tweak the knobs and, finally, we don't see how could it be useful for us so, we kindly thank him/her generosity and time and, we go back home thinking: "what the heck wants he/she that pedal for?. I don't get it!".
To not repeat myself, I would like you to read this previous published entry (in Home Studio section), which will let you understand how a compressor works and what it is useful for:
http://hermeticoguitar.blogspot.com.es/2013/03/home-studio-mixing-part-5.html
T-Rex Comp Nova and MXR Dyna Comp: similar but different
Pedal Effect compressor are more limited in controls than Studio compressors and, in some cases there is just a knob that controls several aspects of the compression: ratio / attack and sustain. In other cases, there is more control over each individual aspect.
Every compressor does basically same function but, none does it in the same way.
T-Rex Comp Nova
The ¨T.Rex Comp Nova is a very transparent compressor. That means that doesn't color the signal and, the signal remain unaffected when the pedal is switched off (true bypass).
It's a modern compressor and, in principle, not based on the design of any of the 3 mythical compressors but, in a new brand design.
It has 3 control knobs: Volume, Comp and Attack.
With the Attack knob, we increase the delay time between the instant were the sound goes over the threshold and the instant were compressor starts its work. More attack, means more delay and, therefore, more transients or peaks so, more punch and natural compression.
With the Comp knob, we determine the compression ratio or, how much the sound that exceeds the threshold will be compressed, once the delay time selected with Attack passed. The more Comp, the more we will drop down peaks and will tame the percussive sound or punch.
The Volume knob is really the Gain or Make Up control of this compressor and, it's responsible to raise the level of the sound in some dB before the compressor does its job so, it determined the output loudness but, also, which part of the sound is over or below the threshold level (fixed inside the compressor).
On its booster mode (Comp rolled off and Attack at max), it has way low gain increase than the MXR unit. Just 3 dB maximum.
For clean stuff, it's lack of special character (colorless) makes it to sound a bit boring, when you compare it against the sound richness that the MXR generates and, therefore, this is where I like it less.
Combined with the Vibe effect, sounds a tad cool and sharp.
Where it shines best is when paired with gain pedals (fuzz, overdrive and distortion), its transparency and smoothness leave a greater dynamic range to a kind of pedals that, by nature, are already compressing the signal (well, clipping the peaks).
Overall, is a friendly pedal, easy to tweak, with a very transparent sound and, this is its greater virtue and defect. That transparency avoid any sonic fingerprint over your own sound and, as a clean booster works well, rising a bit the overall signal and feeding the rest of pedals with a stronger signal.
MXR '74 Vintage Dyna Comp re-issue
It seems that everybody who knew the vintage unit will say that the sound you can achieve with modern versions of the Dyna Comp isn't the same. One of the main reasons is that the new units have a new chip.
Well, it seems that neither this re-issue of this mythical pedal still doesn't sound the same as the original. In comparisons I was able to hear in Youtube, there is clearly a difference in the sound that the reissue edition produces, compared to the original one. The original one is richer in nuances but, in any case, the reissue version sounds way better than the modern Dyna Comp.
So, not having the money and time to source an original Dyna Comp, I went for the reissue version and, well, it has practically all the goods of the original one and all the bad.
It seems that the chip is the same as in the original. They had to source some NOS chips and therefore, this reissues are limited to the number of available chips. So, I bought it before those chips disappear.
Like in the original, there is no plug for an AC transformer so, you can run this pedal just using batteries. You know, if the battery dies, your complete chain of effects disappear (not true bypass).
I made an small hole on the base, to throw there a battery-clip wire (that comes with the Voodoo Lab Power Pedal Plus 2), because if you don't remove the input jack, you battery will dead in short.
As the original it lacks of a led that indicates to you when the pedal is active, so it's a mess when you are working with subtle compression settings. If you end your session with the effect switched on, next time you use your pedalboard the effect will be active without notice (no led).
This pedal has two knobs, highly interdependent. Any change in any of the knobs impacts in the other knob so, any adjust must involve both knobs to achieve the wanted effect. In this sense, the T.Rex is more friendly to achieve the wanted effect.
The Sensitivity knob seems to work over the whole compression character (ratio), while the Output knob controls the Make Up Gain. It seems that the Attack time is fixed inside the unit.
In booster mode, it adds a warm and musical color and has a gain range at least 3 times the range of the T.Rex.
This pedal sounds fantastic with clean stuff (is THE sound) and, perfectly combines with weak effects, as the Vibe or Phaser. Where I don't like it so much is paired with gain pedals (distortion, fuzz), where the compressed signal has a very limited dynamic range.
But used as a booster combined with gain pedals give a nice warm, musical and colorful resulting sound .
The main issue with this pedal is its high noise level. Its gain is so powerful that the existing floor noise is raised in the same amount. Even that you don't notice it so much during playing, the noise clearly pops up when you stop playing and, more specially, with single coils.
Soft compression levels or a "clean" boosting, make the Stratocaster to sound full bodied and with same authority than a Les Paul, maintaining those sweet strato trebles but, with greater richness.
Conclusions
The compressor should be the first pedal in the chain, at least that it enters in conflict with Wah and / or Fuzz. This is the best position because is the spot were the floor noise is the less and, because it modifies the natural dynamics of your guitar, before applying any other effect.
Well, when I bought that MXR reissue, I wasn't very confident of its sound but, it impressed me so good that I've substituted the T.Rex with this MXR.
Anyway, the original issues are still there and, only the time will say if it will remain in my pedalboard.
If you work mainly with clean stuff, the MXR will do a better job but, if you work mainly with gain pedals then, you will love the smoothness and transparency of the T.Rex.
Update Note: After several compressors, my definitive compressor is the Wampler EGO. My search ended there so, be sure to check that one.
Videos
As music is very subjective, I prefer that everyone can see how each compressor works and decide by itself which one better suits its particular needs.
Part 1
In this part, I am describing first the function of a compressor effect and, I start comparing both pedals working with very similar compression levels, to clearly hear how each one colors the signal.
Both pedals are tested with a clean signal and then, with gain pedals (RAT 2 and OCD).
Part 2
In this part, I am comparing both pedals paired with a Vibe and a Fuzz, were we can see very clear differences respect of their work.
Part 3
This third part is exclusively dedicated to explore the possibilities of the MXR Dyna Comp alone.
Part 4
The forth part is dedicated to explore the possibilities of the T.Rex comp Nova.
Everything about electric guitars world
28 April 2013
Amps: Watts and Decibels: a classic misunderstanding
Introduction
Note: this entry was already published in my old Spanish version of this blog around March 2011. I am just revisiting it here for it's interest.
Most of guitarist (and that included myself at some point of time) are confused about Watts and Decibels and, therefore, doesn't understand why some less powered amp can sound louder than a higher powered amp, by example.
The first time anyone hears a Vox AC30 is being impressed by its loudness, it seems louder than any Marshall half stack, by example. While the Vox AC30 is rated at around 30 Watts, typical half or full stacks are rated to 100 Watts.
One more typical confusion is to think that an amp with double power will sound double louder (by example, a 100 W amp = 2 times a 50 W amp).
To understand all this, we need to clarify some concepts, as: Power, Decibels and SPL, among other related ones.
So, let go ahead trying to clarify some things related to amps...
Electrical Power of an Amp (Watts)
Electrical Power definition
Power measures the amount of the Work done by any particular agent in a certain Time.
Imagine that we are a group of friends trying to lift a 1 Kg stone, up to 1 meter high (a table, by example). More or less, everyone will be able to easily lift that stone, in same time.
If we try the same with a 30 Kg stone, to some it will cost a bit more than to others and, the time that will need each one for it will slightly differ from individual to individual.
Now, let try the same with a 80 Kg stone. Probably, most of people couldn't lift that weight up to the table and, those that were able, will need very differentiated times.
For each stone that everybody was able to lift to the same 1 meter high, EVERYONE had done the same amount of WORK (we had to apply same Force to move the stone 1 meter up), everybody burnt the same ENERGY (Energy = Work) but, each person should use a different time for same task.
The quickest is the most powerful, the slowest is the less powerful.
In electricity, power is the product of the electrical potential difference (energy level difference, voltage) and the intensity of the current (amperage) that can be produced by time unit.
Part of such an energy will be dissipated as Heat, most of such an energy will be used for the task which a circuit was designed for.
In the case of guitar amps, the remaining useful energy will be used just to move the coil(s) of your amp' speaker(s).
The Power measurement unit is being called Watt (W).
As any light bulb consumes electrical power from your mains, at a determined ratio / hour, any amp has a power consuming ratio.
The input energy is being transformed in Heat (dissipated by the electronics components of the amp and, specially by tubes, transformers and resistors) and, just part of the original input power remains to let the amp to do its job: to amplify the weak signal of our guitar, converting it into an electrical signal with power enough to move an speaker with ease.
The ratio of the input energy that the amp is able to deliver at its output it's being called: Performance or Efficiency. Given two amps delivering 50 W on their outputs, if the first one consumes 100 W and the second 75 W of mains power, clearly the one consuming 75 W is more efficient than the one consuming 100 W.
The amp consuming 100W was able to use just the 50% of the consumed energy, while the 75W amp was able to use a 66,66% of the consumed energy (or said in other way, wasted just a 33.33% of the consumed energy, against the 50% of the other amp).
Guitar Amps are very inefficient circuits and, waste a lot of energy.
A 50W amp can consume about 125 - 175W to produce just those 50W on its output!.
First surprise for you?.
Power of a tube amp
The power of a tube amp depends, basically, in the efficiency of its tubes and, such an efficiency depends on the tube limitations and on the rest of the amp design.
The so called Single-End designs (one single power tube, by example), are less efficient than the so called Pull-Push designs.
Each tube type can offer a different range of power, up to certain max, depending on the design of the amp.
By example, an EL84 can offer 12W in Single-End pure Class A, 15W in Cathode-Biased and, about 20W in a Pull-Push Class AB design with Fixed-Bias.
Maximum and RMS Power
So, tube types and amp design (Class, Bias type, topology) around those tubes will delimiter the maximum power of such an amp. But, this maximum power will depend on the impedance of the speakers, also.
Usually, an amp is being designed to stand some minimum impedance and several higher impedances.
The less the minimum impedance value is, the harder the amp works.
Typical minimum impedance levels for guitar amps are 4 or 8 Ohms and, can be lower (2 Ohm) in the case of solid state amps for bass guitars.
So, typically, the Power Ratio facilitated by the amp maker is often related the the output power of this amp at its minimum impedance.
If an amp has a maximum power of 100W@4 Ohms and, we plug to it an 8 Ohms cab, we will reduced the amp power approximately to half. Increasing the resistance of the load (impedance), we reduce the voltage and, since Power = voltage * current, delivered power drops.
With a 16 Ohms cab, we would reduce the power to approx a quarter part.
Ok. We can see now that if the cab has a higher impedance than the minimum impedance of the amp, we wouldn't get its maximum power (even that this trick can be used to lower the loudness of the amp).
Therefore, it's very important to read the technical specifications of your amp and, specially in the case of combos, to check which is the minimum impedance for this amp and which impedance have the paired cab.
It's not so rare to find amps with a minimum impedance of 8 Ohms using a paired cab or speaker (in case of combos) rated at 16 Ohm.
This is even more typical in PA systems.
But, things doesn't stop here.
While to measure things that happen in a Direct Current circuit is quite easy, when Alternate Current is involved things go more complex. The AC changes in phase and sign (positive and negative) in a cyclical way.
To approximate values of AC electrical variables to same values measured in DC, a complex mathematical formula was developed. This formula was named Root Mean Square (RMS) and, measures average values.
The amp doesn't delivers its power in a lineal and constant way. The average value between power peaks and valleys is the so called RMS Power and, it's the value that really matters, because it talks about the average power level that the amp can deliver is a constant way.
Peak values should be about 200% over RMS. If the amp is rated to 100W RMS, it's peaks can reach around 200W of maximum power.
Therefore, another important information in amp' specifications is to know if the nominal power is related to Peak or RMS Power. The rule for tube guitar amps is RMS (but, read your specifications).
Then, if our amp delivers 50W RMS, we will need at least an speaker (or cab) supporting 50W RMS (or higher, for higher security) at the same impedance than the amp's minimum impedance.
As amps, speakers rated to certain RMS value can support peaks the double of its RMS value.
So, watch out!. It's not the same a 50W Max amp and a 50W RMS amp. The 50W Max amp will have 50W peaks but, will deliver around 25W RMS, while the 50W RMS will have 100W peaks and will deliver 50W RMS (double power).
Well, mix this Max / RMS thing with impedance mismatching and, you can start to understand how a theoretically powerful amp (75W Max) can sound quieter than a theoretically less powerful amp (50W RMS).
But, we are still missing the other big component: Speakers.
Speakers
At this point, it's clear that all that output energy (power) that the amp is delivering will be used just to move the big coil of our speaker(s), to move the membrane, to push the air (by pressuring it) and, creating the sound but... which role play speakers in all this confusion about loudness and power?.
In one side, we already discussed about the fact that the speaker impedance increases the resistance in the power stage of the amp, dropping the delivered power (and thus, the available energy that can be transduced into sound). But, neither the Watts of Power of an amp or an speaker tell us nothing about their own efficiency in to transform that available input energy (output of the amp, input of the speaker) to sound.
As in the case of amps, the speaker will waste part of its input energy (dissipated as heat) and use the remaining one to excite the coil to move its membrane. But, we need to clarify still some things, before going further.
Decibels (dB)
The Decibel is the 10th part of a Bel. A Bel establishes the logarithmic ratio for the power difference of a certain value related to a reference value.
In the case of sound, we arbitrary assigned a zero value to the threshold of human hearing (that is, 0 dB = threshold of human hearing).
This is so arbitrary and useful as to give a zero value to the level of the sea and then, to measure mountains, expressing their high related to that reference value.
It's important to say here that the Decibel, in fact, is an undimensional unit and, means nothing alone, at least that is being related to some other dimensional unit.
Together with the decibel symbol, we will need to add some dimensional symbol to properly define what are we measuring.
So, we can see 1 dB V (voltage) , 1 dB SPL (sound pressure level), 1 dB RMS (average level), etc.
As it's a logarithmic unit, each increase of 1 Bel means a power 10 times higher respect of the previous power level so, 1 dB is 10 times more powerful that 0 dB and, 2 dB 10 times more powerful than 1 dB and 100 times more powerful than 0 dB.
Since the Bel is a huge measurement unit, the decibel (a 10th of a Bel) is being used to measure loudness, instead. The Pain Threshold is about 140 dB (respect of the hearing threshold!).
To have a better idea, a quiet breathing sounds around 10 dB, a conversation around 40 dB, traffic noise around 90 dB, 98 dB is the standard for Cinema, 130 dB correspond to a reaction plane taking off and, the biggest measured sound level ever measured were the 180 dB produced by the explosion of Krakatoa Vulcan.
Sound Pressure Level (SPL)
The Sound Pressure Level measures the instantaneous intensity of the sound that reaches a certain spot.
Even that air pressure changes are being measured in Pascals, since there is a really wide range between the hearing threshold (20 micro-Pascals) and the pain threshold (200 Pascals), we use the decibel scale (logarithmic), taking as the reference value (0 dB SPL) the 20 micro-Pascals that correspond to the the hearing threshold.
Usually, speaker specifications say which are the maximum SPL that they can generate and, such a level is often referred to the level perceived at 1 meter of the speaker.
Since the sound is being produced when compressing the air molecules, that travel in the space as waves, pressuring our eardrum (inversely to how an speaker works), loudness changes are being measured in decibels SPL and, not in Watts. That's a common confusion.
There is some relationship between Watts and dB SPL but, this is not a 1 to 1 relationship. If we double the watts (by example, from 50W RMS to 100W RMS), we are increasing the sound pressure in about 3 dB (not doubling it!). If, by example, 50W produced 102 dB SPL, 100W will produce 105 dB SPL and not 204 dB SPL (remember that the loudest sound never recorded reached "just" 180 dB!!!).
Speaker efficiency
In the same way that amp are differently efficient transforming the input power into output power, not every speaker is equally efficient transducing the electrical power into sound pressure (sound).
In principle, speakers that support higher power levels should be able to move more air than speakers of lower power and, this is like this but, when comparing several speakers with equal power, not all them offer same efficiency levels (sound pressure).
By example, the Celestion G12 "Blue Bulldog", that were originally mounted in those Vox AC30 were able to deliver 100 dB SPL, while the Jensen of same epoch (Fender amps, basically) were generating between 90 and 96 dB SPL. The famous Celestion Greenback deliver 97 dB SPL, by example.
3 dB SPL more or less is the minimum difference that the common human hears as an stepped change in loudness. Remember that this is the difference between the loudness of a 100W amp and a 50W amp or, between a 50W amp and a 25W amp.
Speakers cab design
Size and geometry of the own speaker (membrane, coil, etc.) determines the amount of air that the speaker can push in a single stroke (movement of the membrane). Two speakers move more air than just one and, 4 move more than 2 and, 8 more than 4. That's clear.
Together with the number of speakers and their own efficiency, the design of the cab (dimensions, shape, components...) that holds them, contributes to how the sound is actually being projected and, also to which frequencies are more or less represented (what is very important, since the human hearing is more sensible to mid frequencies, the closer to the human voices).
Big Ass, walking or don't walking?
So nice so good. We have already seen that there are several factors that will determine if a certain amp will sound louder than other one and, this is not always directly proportional to their nominal power so, everything depends on the efficiency of the set amp-speakers.
So, ok, I've got it: I will pair the speakers with the higher efficiency with the impedance level corresponding to the minimum impedance supported by my amp and... krank that shit!.
Errr.... NOT. This is not a competition to see who has the bigger thing. We are talking about music and, music is being perceived in a very subjective way.
The set amp-speakers is our main tool as guitarists (wasn't it the guitar?... then not... even a cheap guitar can sound nice in a great amp, the opposite is false) and, there are amps and speakers to cover every taste.
When an amplifier is being designed, the maker is also choosing certain speakers to achieve a certain sound. If that sound it's being achieved with the less efficient speaker then, this is the right speaker and, it doesn't matter if the amp of our neighbor sounds way louder, if it doesn't suits our needs.
As I find stupid to drive a Ferrari Testarrosa in the city (well, you look really cool and will attract some girls, this I can recognize but... can you really enjoy the potential of such a car?), it's absolutely stupid to have a Full Stack of 100W with two cabs of 4x12" speakers (and Blue Bulldog, why not), to play at 70 dB SPL (bellow traffic noise).
We will not extract the best of that amp and speakers, because both will be work way bellow their sweet spots, where they can deliver their magic voice.
Therefore, you will see often en Recording Studios (by example) little amps that you will be able to krank to their sweet spot to record harmonically rich guitars. Most of the tracks we love were recorded with little amps. Some samples are Fender Tweed, Champ, Blues Jr and similar.
The best disk of Clapton was recorded with a 18W RMS combo (called later BluesBreaker, because of the disk). In most of Zeppelin disks, guitars were recorded using a Supro amp rated at 15-20W RMS.
Clearly, you don't need same sound pressure to fill a stadium or to fill an small room.
The power and efficiency of your amp and speakers has to be the right one for the space and loudness that you need for each environment.
A side warning note about speaker cables
Unfortunately, I've seen many guitarists and, even some "sound engineer", using guitar cables to link the cab to the amp. This is definitively wrong.
Anyone that knows a bit about amps (and, I am not talking about myself, since I know nothing about), will say to you that you have to use speaker cables, otherwise you are in risk to damage your speaker and / or your amp (usually the Output Transformer is what deads here, costing an eye to repair).
You should had hear it but, nobody explained to you why, am I right? (and that's why you think it's bullshit).
We have seen above that the output power of an amp is the product of voltage and current. If the output voltage is 370 Volts and the amp is rated to 50W RMS, that means that the current value is:
P = V * I, then I = P / V
I = 50 W / 370 V = 0,135 A = 135 mA
In instrument cables, concretely guitar cables, travel voltage levels in the order of mili-volts (about 1000 times lower) and, very low current levels and, therefore, the energy travelling the cable is way lower (about 1000 times lower).
Depending on the power expected, the cable should have a different section (gauge).
Signals of low power can travel producing none or very low heating in conductors of a gauge similar to a hair but, higher power signals will need wider sections, to allow the conductor to "eat" electrons easily, and reducing the heat generated because of the resistance that the conductor offers to the travel of electrons.
Look at your mains power.
From the Energy Central there exit cables of a huge gauge, travelling all the country.
From those cables, smaller gauged cables reach your house's entry spot.
Then, from that entry spot, smaller gauged cables distribute the energy inside your home.
Your electric network will be protected by a therm-magnetic device that will jump if there is some overheating in your cabling.
This overheating can occur because your are hanging one or more electrical devices in the same line which, combined power summed up together is bigger than the power that the conductor can stand.
If that protective device wasn't there and the overheat stant for a long while, it will produce fire.
For this same reason, the cable that it's used to link the amp and the speaker or cab, must be a cable with TWO independent (positive and negative) conductors, of an adequate section.
Instrument cables have just a single signal conductor, wrapped in a mall or a metallic folder to ground such a signal. Those cable haven't the right section to support the power generated in the amp.
Therefore, the overheating of such a cables can even degenerate in a fire and, disappearing the load of the speaker, your amp can result seriously damaged (and even get fire).
So, please, buy some speaker cable for your amp and cab and, forget using instrument cables for that.
You never know when the overheat will be enough to cause the issue.
Note: this entry was already published in my old Spanish version of this blog around March 2011. I am just revisiting it here for it's interest.
Most of guitarist (and that included myself at some point of time) are confused about Watts and Decibels and, therefore, doesn't understand why some less powered amp can sound louder than a higher powered amp, by example.
The first time anyone hears a Vox AC30 is being impressed by its loudness, it seems louder than any Marshall half stack, by example. While the Vox AC30 is rated at around 30 Watts, typical half or full stacks are rated to 100 Watts.
One more typical confusion is to think that an amp with double power will sound double louder (by example, a 100 W amp = 2 times a 50 W amp).
To understand all this, we need to clarify some concepts, as: Power, Decibels and SPL, among other related ones.
So, let go ahead trying to clarify some things related to amps...
Electrical Power of an Amp (Watts)
Electrical Power definition
Power measures the amount of the Work done by any particular agent in a certain Time.
Imagine that we are a group of friends trying to lift a 1 Kg stone, up to 1 meter high (a table, by example). More or less, everyone will be able to easily lift that stone, in same time.
If we try the same with a 30 Kg stone, to some it will cost a bit more than to others and, the time that will need each one for it will slightly differ from individual to individual.
Now, let try the same with a 80 Kg stone. Probably, most of people couldn't lift that weight up to the table and, those that were able, will need very differentiated times.
For each stone that everybody was able to lift to the same 1 meter high, EVERYONE had done the same amount of WORK (we had to apply same Force to move the stone 1 meter up), everybody burnt the same ENERGY (Energy = Work) but, each person should use a different time for same task.
The quickest is the most powerful, the slowest is the less powerful.
In electricity, power is the product of the electrical potential difference (energy level difference, voltage) and the intensity of the current (amperage) that can be produced by time unit.
Part of such an energy will be dissipated as Heat, most of such an energy will be used for the task which a circuit was designed for.
In the case of guitar amps, the remaining useful energy will be used just to move the coil(s) of your amp' speaker(s).
The Power measurement unit is being called Watt (W).
As any light bulb consumes electrical power from your mains, at a determined ratio / hour, any amp has a power consuming ratio.
The input energy is being transformed in Heat (dissipated by the electronics components of the amp and, specially by tubes, transformers and resistors) and, just part of the original input power remains to let the amp to do its job: to amplify the weak signal of our guitar, converting it into an electrical signal with power enough to move an speaker with ease.
The ratio of the input energy that the amp is able to deliver at its output it's being called: Performance or Efficiency. Given two amps delivering 50 W on their outputs, if the first one consumes 100 W and the second 75 W of mains power, clearly the one consuming 75 W is more efficient than the one consuming 100 W.
The amp consuming 100W was able to use just the 50% of the consumed energy, while the 75W amp was able to use a 66,66% of the consumed energy (or said in other way, wasted just a 33.33% of the consumed energy, against the 50% of the other amp).
Guitar Amps are very inefficient circuits and, waste a lot of energy.
A 50W amp can consume about 125 - 175W to produce just those 50W on its output!.
First surprise for you?.
Power of a tube amp
The power of a tube amp depends, basically, in the efficiency of its tubes and, such an efficiency depends on the tube limitations and on the rest of the amp design.
The so called Single-End designs (one single power tube, by example), are less efficient than the so called Pull-Push designs.
Each tube type can offer a different range of power, up to certain max, depending on the design of the amp.
By example, an EL84 can offer 12W in Single-End pure Class A, 15W in Cathode-Biased and, about 20W in a Pull-Push Class AB design with Fixed-Bias.
Maximum and RMS Power
So, tube types and amp design (Class, Bias type, topology) around those tubes will delimiter the maximum power of such an amp. But, this maximum power will depend on the impedance of the speakers, also.
Usually, an amp is being designed to stand some minimum impedance and several higher impedances.
The less the minimum impedance value is, the harder the amp works.
Typical minimum impedance levels for guitar amps are 4 or 8 Ohms and, can be lower (2 Ohm) in the case of solid state amps for bass guitars.
So, typically, the Power Ratio facilitated by the amp maker is often related the the output power of this amp at its minimum impedance.
If an amp has a maximum power of 100W@4 Ohms and, we plug to it an 8 Ohms cab, we will reduced the amp power approximately to half. Increasing the resistance of the load (impedance), we reduce the voltage and, since Power = voltage * current, delivered power drops.
With a 16 Ohms cab, we would reduce the power to approx a quarter part.
Ok. We can see now that if the cab has a higher impedance than the minimum impedance of the amp, we wouldn't get its maximum power (even that this trick can be used to lower the loudness of the amp).
Therefore, it's very important to read the technical specifications of your amp and, specially in the case of combos, to check which is the minimum impedance for this amp and which impedance have the paired cab.
It's not so rare to find amps with a minimum impedance of 8 Ohms using a paired cab or speaker (in case of combos) rated at 16 Ohm.
This is even more typical in PA systems.
But, things doesn't stop here.
While to measure things that happen in a Direct Current circuit is quite easy, when Alternate Current is involved things go more complex. The AC changes in phase and sign (positive and negative) in a cyclical way.
To approximate values of AC electrical variables to same values measured in DC, a complex mathematical formula was developed. This formula was named Root Mean Square (RMS) and, measures average values.
The amp doesn't delivers its power in a lineal and constant way. The average value between power peaks and valleys is the so called RMS Power and, it's the value that really matters, because it talks about the average power level that the amp can deliver is a constant way.
Peak values should be about 200% over RMS. If the amp is rated to 100W RMS, it's peaks can reach around 200W of maximum power.
Therefore, another important information in amp' specifications is to know if the nominal power is related to Peak or RMS Power. The rule for tube guitar amps is RMS (but, read your specifications).
Then, if our amp delivers 50W RMS, we will need at least an speaker (or cab) supporting 50W RMS (or higher, for higher security) at the same impedance than the amp's minimum impedance.
As amps, speakers rated to certain RMS value can support peaks the double of its RMS value.
So, watch out!. It's not the same a 50W Max amp and a 50W RMS amp. The 50W Max amp will have 50W peaks but, will deliver around 25W RMS, while the 50W RMS will have 100W peaks and will deliver 50W RMS (double power).
Well, mix this Max / RMS thing with impedance mismatching and, you can start to understand how a theoretically powerful amp (75W Max) can sound quieter than a theoretically less powerful amp (50W RMS).
But, we are still missing the other big component: Speakers.
Speakers
At this point, it's clear that all that output energy (power) that the amp is delivering will be used just to move the big coil of our speaker(s), to move the membrane, to push the air (by pressuring it) and, creating the sound but... which role play speakers in all this confusion about loudness and power?.
In one side, we already discussed about the fact that the speaker impedance increases the resistance in the power stage of the amp, dropping the delivered power (and thus, the available energy that can be transduced into sound). But, neither the Watts of Power of an amp or an speaker tell us nothing about their own efficiency in to transform that available input energy (output of the amp, input of the speaker) to sound.
As in the case of amps, the speaker will waste part of its input energy (dissipated as heat) and use the remaining one to excite the coil to move its membrane. But, we need to clarify still some things, before going further.
Decibels (dB)
The Decibel is the 10th part of a Bel. A Bel establishes the logarithmic ratio for the power difference of a certain value related to a reference value.
In the case of sound, we arbitrary assigned a zero value to the threshold of human hearing (that is, 0 dB = threshold of human hearing).
This is so arbitrary and useful as to give a zero value to the level of the sea and then, to measure mountains, expressing their high related to that reference value.
It's important to say here that the Decibel, in fact, is an undimensional unit and, means nothing alone, at least that is being related to some other dimensional unit.
Together with the decibel symbol, we will need to add some dimensional symbol to properly define what are we measuring.
So, we can see 1 dB V (voltage) , 1 dB SPL (sound pressure level), 1 dB RMS (average level), etc.
As it's a logarithmic unit, each increase of 1 Bel means a power 10 times higher respect of the previous power level so, 1 dB is 10 times more powerful that 0 dB and, 2 dB 10 times more powerful than 1 dB and 100 times more powerful than 0 dB.
Since the Bel is a huge measurement unit, the decibel (a 10th of a Bel) is being used to measure loudness, instead. The Pain Threshold is about 140 dB (respect of the hearing threshold!).
To have a better idea, a quiet breathing sounds around 10 dB, a conversation around 40 dB, traffic noise around 90 dB, 98 dB is the standard for Cinema, 130 dB correspond to a reaction plane taking off and, the biggest measured sound level ever measured were the 180 dB produced by the explosion of Krakatoa Vulcan.
Sound Pressure Level (SPL)
The Sound Pressure Level measures the instantaneous intensity of the sound that reaches a certain spot.
Even that air pressure changes are being measured in Pascals, since there is a really wide range between the hearing threshold (20 micro-Pascals) and the pain threshold (200 Pascals), we use the decibel scale (logarithmic), taking as the reference value (0 dB SPL) the 20 micro-Pascals that correspond to the the hearing threshold.
Usually, speaker specifications say which are the maximum SPL that they can generate and, such a level is often referred to the level perceived at 1 meter of the speaker.
Since the sound is being produced when compressing the air molecules, that travel in the space as waves, pressuring our eardrum (inversely to how an speaker works), loudness changes are being measured in decibels SPL and, not in Watts. That's a common confusion.
There is some relationship between Watts and dB SPL but, this is not a 1 to 1 relationship. If we double the watts (by example, from 50W RMS to 100W RMS), we are increasing the sound pressure in about 3 dB (not doubling it!). If, by example, 50W produced 102 dB SPL, 100W will produce 105 dB SPL and not 204 dB SPL (remember that the loudest sound never recorded reached "just" 180 dB!!!).
Speaker efficiency
In the same way that amp are differently efficient transforming the input power into output power, not every speaker is equally efficient transducing the electrical power into sound pressure (sound).
In principle, speakers that support higher power levels should be able to move more air than speakers of lower power and, this is like this but, when comparing several speakers with equal power, not all them offer same efficiency levels (sound pressure).
By example, the Celestion G12 "Blue Bulldog", that were originally mounted in those Vox AC30 were able to deliver 100 dB SPL, while the Jensen of same epoch (Fender amps, basically) were generating between 90 and 96 dB SPL. The famous Celestion Greenback deliver 97 dB SPL, by example.
3 dB SPL more or less is the minimum difference that the common human hears as an stepped change in loudness. Remember that this is the difference between the loudness of a 100W amp and a 50W amp or, between a 50W amp and a 25W amp.
Speakers cab design
Size and geometry of the own speaker (membrane, coil, etc.) determines the amount of air that the speaker can push in a single stroke (movement of the membrane). Two speakers move more air than just one and, 4 move more than 2 and, 8 more than 4. That's clear.
Together with the number of speakers and their own efficiency, the design of the cab (dimensions, shape, components...) that holds them, contributes to how the sound is actually being projected and, also to which frequencies are more or less represented (what is very important, since the human hearing is more sensible to mid frequencies, the closer to the human voices).
Big Ass, walking or don't walking?
So nice so good. We have already seen that there are several factors that will determine if a certain amp will sound louder than other one and, this is not always directly proportional to their nominal power so, everything depends on the efficiency of the set amp-speakers.
So, ok, I've got it: I will pair the speakers with the higher efficiency with the impedance level corresponding to the minimum impedance supported by my amp and... krank that shit!.
Errr.... NOT. This is not a competition to see who has the bigger thing. We are talking about music and, music is being perceived in a very subjective way.
The set amp-speakers is our main tool as guitarists (wasn't it the guitar?... then not... even a cheap guitar can sound nice in a great amp, the opposite is false) and, there are amps and speakers to cover every taste.
When an amplifier is being designed, the maker is also choosing certain speakers to achieve a certain sound. If that sound it's being achieved with the less efficient speaker then, this is the right speaker and, it doesn't matter if the amp of our neighbor sounds way louder, if it doesn't suits our needs.
As I find stupid to drive a Ferrari Testarrosa in the city (well, you look really cool and will attract some girls, this I can recognize but... can you really enjoy the potential of such a car?), it's absolutely stupid to have a Full Stack of 100W with two cabs of 4x12" speakers (and Blue Bulldog, why not), to play at 70 dB SPL (bellow traffic noise).
We will not extract the best of that amp and speakers, because both will be work way bellow their sweet spots, where they can deliver their magic voice.
Therefore, you will see often en Recording Studios (by example) little amps that you will be able to krank to their sweet spot to record harmonically rich guitars. Most of the tracks we love were recorded with little amps. Some samples are Fender Tweed, Champ, Blues Jr and similar.
The best disk of Clapton was recorded with a 18W RMS combo (called later BluesBreaker, because of the disk). In most of Zeppelin disks, guitars were recorded using a Supro amp rated at 15-20W RMS.
Clearly, you don't need same sound pressure to fill a stadium or to fill an small room.
The power and efficiency of your amp and speakers has to be the right one for the space and loudness that you need for each environment.
A side warning note about speaker cables
Unfortunately, I've seen many guitarists and, even some "sound engineer", using guitar cables to link the cab to the amp. This is definitively wrong.
Anyone that knows a bit about amps (and, I am not talking about myself, since I know nothing about), will say to you that you have to use speaker cables, otherwise you are in risk to damage your speaker and / or your amp (usually the Output Transformer is what deads here, costing an eye to repair).
You should had hear it but, nobody explained to you why, am I right? (and that's why you think it's bullshit).
We have seen above that the output power of an amp is the product of voltage and current. If the output voltage is 370 Volts and the amp is rated to 50W RMS, that means that the current value is:
P = V * I, then I = P / V
I = 50 W / 370 V = 0,135 A = 135 mA
In instrument cables, concretely guitar cables, travel voltage levels in the order of mili-volts (about 1000 times lower) and, very low current levels and, therefore, the energy travelling the cable is way lower (about 1000 times lower).
Depending on the power expected, the cable should have a different section (gauge).
Signals of low power can travel producing none or very low heating in conductors of a gauge similar to a hair but, higher power signals will need wider sections, to allow the conductor to "eat" electrons easily, and reducing the heat generated because of the resistance that the conductor offers to the travel of electrons.
Look at your mains power.
From the Energy Central there exit cables of a huge gauge, travelling all the country.
From those cables, smaller gauged cables reach your house's entry spot.
Then, from that entry spot, smaller gauged cables distribute the energy inside your home.
Your electric network will be protected by a therm-magnetic device that will jump if there is some overheating in your cabling.
This overheating can occur because your are hanging one or more electrical devices in the same line which, combined power summed up together is bigger than the power that the conductor can stand.
If that protective device wasn't there and the overheat stant for a long while, it will produce fire.
For this same reason, the cable that it's used to link the amp and the speaker or cab, must be a cable with TWO independent (positive and negative) conductors, of an adequate section.
Instrument cables have just a single signal conductor, wrapped in a mall or a metallic folder to ground such a signal. Those cable haven't the right section to support the power generated in the amp.
Therefore, the overheating of such a cables can even degenerate in a fire and, disappearing the load of the speaker, your amp can result seriously damaged (and even get fire).
So, please, buy some speaker cable for your amp and cab and, forget using instrument cables for that.
You never know when the overheat will be enough to cause the issue.
17 April 2013
Amps: Harley Benton 4x12" Vintage Cab
Introduction
Note: this entry was already published in my old Spanish version of this blog, around March/2011. I am revisiting it here.
A friend of mine received as a gift a Marshall Valvestate 8200 Bi-Chorus head, an hybrid amp that can work in stereo, delivery 100 W RMS by channel.
This bad boy was used by people like Panthera and, needs to have a speaker cab connected to each channel, otherwise can be fried.
My friend "stores" his cab in my home and, I hadn't any cab able to handle that amount of watts so, I've started to search for some valid cab to test the amp head.
Immediately, the Marshall 1960A comes to mind but, the price was higher enough just to test the head so, I browsed again, looking for a more affordable alternative.
I was gratefully surprised with the price of the Harley Benton 4x12 Vintage cab, it was mounting 4 Celestion Vintage 30 (V30), maybe not the best option for a Metal amp but, a bargain, anyway.
Well, the cab is here and, I was testing it so, what follows are my own impressions.
Presentation
Harley Benton is the white brand by Thomann. Previously, I gave a Harley Benton strato-like guitar to my brother in law and, I was surprised for its quality, taking into account the cheap price. Honestly, I've seen expensive guitars worst finished. The wood of that guitar was ok and well finished, despite of the cheap hardware and electronics.
So, I've decided to give a try to that Harley Benton cab, at the end, it was mounting 4 Celestion V30 that could be mounted in any other cab, if things were wrong.
Before buying it, I was googling a while and, I saw some people that already bought it and, that had the patience to open the cab and look inside. The cab was made of plywood, instead of the cheap DM material that many other cheap cabs are made of (as Framus cabs, by example).
The external look, as well as dimensions and wall thickness of such a cab were very similar to those of Marshall's and, they mounted true V30s.
So... I didn't saw any reason to not buy that cab.
Being a bulky good, the cab came palletized in a truck, differently of the typical order that comes via UPS, by example.
It took me a while to unwrap the cab. They had put the casters just in the hole of the handles so, to get the handles on hands to support the cab and while kicking off the cardboard box was a bit difficult, but... finally I've success.
Casters, instead of being inserted by pushing them, go screwed and, once in place, they allow to move the cab with ease.
Before plugging the cab, I had a look to the overall aspect of the cab and, it pleased me. The finishing is perfect, same quality you could found in a good cab. Squares had some metallic reinforcement, painted in matte black, narrower that the ones in Marshall's but, useful anyway.
The handles, the tolex... everything was ok.
Satisfied with the external look, it was time to check the cab.
Breaking in speakers
First task, when you have a new cab or speaker is to break them in. Speakers are adapting themselves during their life and, they aren't usually delivering their best sound at the beginning.
According to Celestion, there is a "breaking in" process that allows us to achieve the 95% of expected sound, after a short breaking in time. The remaining 5% could take a long time. In fact, in my other amps, speakers seem to have being finally settled after a year, more or less.
To start the breaking in process, you need to work with the clean channel, with all your tone stack at noon, low gain and low volume, for about 15 mins. You should leave the speakers to be warming up for this time, playing softly and variate, at low volume.
Once the warming up process finishes, you have to put gain at minimum and volume at full. Mids and Bass at full and, Treble at least at noon. Gain will be increased step by step during this second phase, until to reach a good loudness but, without being painful.
You need to strum strongly strings, trying to achieve peaks in basses, mids and trebles. Power chords force the coil to be moved in all directions and with all intensities what will help to settle the cone.
This should take 15 minutes more.
So, after half hour, your speakers should be delivering the 95% of their best sound.
The Sound
And, that's the key question... do they sound nice?.
ABSOLUTELY!.
Geeks will say that being so cheap, those speakers have been made in China (honestly, what else don't?). Look at around you, your mobile, your PC motherboard, your TV...
This cab mounts authentic V30 and, they sound as they should, wherever they were made.
I cannot understand how Thomann can offer such a kind of cab to a price that is just slightly over the price of a single speaker. I dunno how but, if you are searching for a 4x12" V30 cab, be sure to check this one.
At the beginning, we were trying to test the Valvestate cab of my friend but, unsuccessfully. The head seems to be out-of-order. I've even swapped the original tube with a new brand one.
Then, I've tried the Orange Rockerverb 50 for next test and, this is the amp I've used to perform the breaking in process. After the breaking in, I've started to check several guitars plugged to the Orange's and, I liked a lot what I've heard.
Then, I've switched off the Rockerveb 50 and, switched on the Vox Night Train. Being the smallest of my amps, I was curious about how it will sound thru a 4x12 cab.
Atomic Mother! What a craziness!. With that cab, the NT is a beast. I wasn't able to raise the volume beyond 11:00, without starting to suffer pain in my ears.
The LP sounded to death, as the Charvel did but... the Strato... Pink Floyd Live!.
Tomorrow or, during next weekend, I would like to test it with the Koch Studiotone and the Marshall 1923C (with which I am expecting the most spectacular results, thus the last I want to test).
If I have some time and willing, I will prepare some video with part of the tests.
Since speakers aren't totally settled, it seemed to me as if they had an "slow" response at the beginning, something that went to better while playing. At the very beginning, it seemed as if the notes were "glued" to the cone, making it difficult to play speedy riffs.
One more sensation I had the very first day is that those speakers break up really early. In my rest of speakers, I can achieve cleaner sounds at higher volumes / gain levels while, I've been forced to lower the gain and/or volume to achieve a clean sound with this cab. Otherwise, it's really easy to break up the speaker and enter in Blues or Hard Rock territory.
I suppose that the way that the amp interacts with that 4x12" cab is totally different to how it does it with the stock speakers but, ok, to achieve the target sound is just a matter of re-tweak your amp's knobs, at the end.
What is impressive is the amount of air that this cab moves. I've maintained low level in each amp and, the sound was slapping me hard. I finished half deaf.
This cab needs some room around, since it produces a true sonic Tsunami. I think even my name was vibrating, while checking the gear. I cannot imagine how can kicks you with a 1959 Super Lead at full.
Video
I've prepared a video where I am firstly showing the finishing details of the cab and, then a demo of its sounds, using as a basis a free downloadable version of "Shine on You crazy diamonds" by Pink Floyd.
Just focus on the sound and, as ever, forget my playing.
Conclusion
A 4x12" cab, with 4 authentic Celestion V30, made of plywood, with similar building characteristics as a Marshall 1960A cab and, at a price that is just slightly expensive than a single V30 speaker?.
Don't doubt it, it's a bargain!.
If you are looking for a 4x12" cab, with a Vintage sound and, you don't check this one, you are loosing a great opportunity, in my honest opinion.
The only drawback I can imagine is its potential re-sale value, because its logo doesn't corresponds to a big brand. But, this negative point for the seller will be a godsend for the buyer.
Note: this entry was already published in my old Spanish version of this blog, around March/2011. I am revisiting it here.
A friend of mine received as a gift a Marshall Valvestate 8200 Bi-Chorus head, an hybrid amp that can work in stereo, delivery 100 W RMS by channel.
This bad boy was used by people like Panthera and, needs to have a speaker cab connected to each channel, otherwise can be fried.
My friend "stores" his cab in my home and, I hadn't any cab able to handle that amount of watts so, I've started to search for some valid cab to test the amp head.
Immediately, the Marshall 1960A comes to mind but, the price was higher enough just to test the head so, I browsed again, looking for a more affordable alternative.
I was gratefully surprised with the price of the Harley Benton 4x12 Vintage cab, it was mounting 4 Celestion Vintage 30 (V30), maybe not the best option for a Metal amp but, a bargain, anyway.
Well, the cab is here and, I was testing it so, what follows are my own impressions.
Presentation
Harley Benton is the white brand by Thomann. Previously, I gave a Harley Benton strato-like guitar to my brother in law and, I was surprised for its quality, taking into account the cheap price. Honestly, I've seen expensive guitars worst finished. The wood of that guitar was ok and well finished, despite of the cheap hardware and electronics.
So, I've decided to give a try to that Harley Benton cab, at the end, it was mounting 4 Celestion V30 that could be mounted in any other cab, if things were wrong.
Before buying it, I was googling a while and, I saw some people that already bought it and, that had the patience to open the cab and look inside. The cab was made of plywood, instead of the cheap DM material that many other cheap cabs are made of (as Framus cabs, by example).
The external look, as well as dimensions and wall thickness of such a cab were very similar to those of Marshall's and, they mounted true V30s.
So... I didn't saw any reason to not buy that cab.
Being a bulky good, the cab came palletized in a truck, differently of the typical order that comes via UPS, by example.
It took me a while to unwrap the cab. They had put the casters just in the hole of the handles so, to get the handles on hands to support the cab and while kicking off the cardboard box was a bit difficult, but... finally I've success.
Casters, instead of being inserted by pushing them, go screwed and, once in place, they allow to move the cab with ease.
Before plugging the cab, I had a look to the overall aspect of the cab and, it pleased me. The finishing is perfect, same quality you could found in a good cab. Squares had some metallic reinforcement, painted in matte black, narrower that the ones in Marshall's but, useful anyway.
The handles, the tolex... everything was ok.
Satisfied with the external look, it was time to check the cab.
Breaking in speakers
First task, when you have a new cab or speaker is to break them in. Speakers are adapting themselves during their life and, they aren't usually delivering their best sound at the beginning.
According to Celestion, there is a "breaking in" process that allows us to achieve the 95% of expected sound, after a short breaking in time. The remaining 5% could take a long time. In fact, in my other amps, speakers seem to have being finally settled after a year, more or less.
To start the breaking in process, you need to work with the clean channel, with all your tone stack at noon, low gain and low volume, for about 15 mins. You should leave the speakers to be warming up for this time, playing softly and variate, at low volume.
Once the warming up process finishes, you have to put gain at minimum and volume at full. Mids and Bass at full and, Treble at least at noon. Gain will be increased step by step during this second phase, until to reach a good loudness but, without being painful.
You need to strum strongly strings, trying to achieve peaks in basses, mids and trebles. Power chords force the coil to be moved in all directions and with all intensities what will help to settle the cone.
This should take 15 minutes more.
So, after half hour, your speakers should be delivering the 95% of their best sound.
The Sound
And, that's the key question... do they sound nice?.
ABSOLUTELY!.
Geeks will say that being so cheap, those speakers have been made in China (honestly, what else don't?). Look at around you, your mobile, your PC motherboard, your TV...
This cab mounts authentic V30 and, they sound as they should, wherever they were made.
I cannot understand how Thomann can offer such a kind of cab to a price that is just slightly over the price of a single speaker. I dunno how but, if you are searching for a 4x12" V30 cab, be sure to check this one.
At the beginning, we were trying to test the Valvestate cab of my friend but, unsuccessfully. The head seems to be out-of-order. I've even swapped the original tube with a new brand one.
Then, I've tried the Orange Rockerverb 50 for next test and, this is the amp I've used to perform the breaking in process. After the breaking in, I've started to check several guitars plugged to the Orange's and, I liked a lot what I've heard.
Then, I've switched off the Rockerveb 50 and, switched on the Vox Night Train. Being the smallest of my amps, I was curious about how it will sound thru a 4x12 cab.
Atomic Mother! What a craziness!. With that cab, the NT is a beast. I wasn't able to raise the volume beyond 11:00, without starting to suffer pain in my ears.
The LP sounded to death, as the Charvel did but... the Strato... Pink Floyd Live!.
Tomorrow or, during next weekend, I would like to test it with the Koch Studiotone and the Marshall 1923C (with which I am expecting the most spectacular results, thus the last I want to test).
If I have some time and willing, I will prepare some video with part of the tests.
Since speakers aren't totally settled, it seemed to me as if they had an "slow" response at the beginning, something that went to better while playing. At the very beginning, it seemed as if the notes were "glued" to the cone, making it difficult to play speedy riffs.
One more sensation I had the very first day is that those speakers break up really early. In my rest of speakers, I can achieve cleaner sounds at higher volumes / gain levels while, I've been forced to lower the gain and/or volume to achieve a clean sound with this cab. Otherwise, it's really easy to break up the speaker and enter in Blues or Hard Rock territory.
I suppose that the way that the amp interacts with that 4x12" cab is totally different to how it does it with the stock speakers but, ok, to achieve the target sound is just a matter of re-tweak your amp's knobs, at the end.
What is impressive is the amount of air that this cab moves. I've maintained low level in each amp and, the sound was slapping me hard. I finished half deaf.
This cab needs some room around, since it produces a true sonic Tsunami. I think even my name was vibrating, while checking the gear. I cannot imagine how can kicks you with a 1959 Super Lead at full.
Video
I've prepared a video where I am firstly showing the finishing details of the cab and, then a demo of its sounds, using as a basis a free downloadable version of "Shine on You crazy diamonds" by Pink Floyd.
Just focus on the sound and, as ever, forget my playing.
Conclusion
A 4x12" cab, with 4 authentic Celestion V30, made of plywood, with similar building characteristics as a Marshall 1960A cab and, at a price that is just slightly expensive than a single V30 speaker?.
Don't doubt it, it's a bargain!.
If you are looking for a 4x12" cab, with a Vintage sound and, you don't check this one, you are loosing a great opportunity, in my honest opinion.
The only drawback I can imagine is its potential re-sale value, because its logo doesn't corresponds to a big brand. But, this negative point for the seller will be a godsend for the buyer.
16 April 2013
Accessories: Yamaha StagePass 300 PA System
Introduction
Note: this entry was already published in my old Spanish blog, during February/2012. I am just revisiting it with added info, based on the experience more than one year later.
To practice with backing tracks is interesting to me for several reasons. In first place, I avoid to go boring playing the guitar alone without any particular target. In second place, this helps me to "couple" my sound and rig to a "virtual band of musics".
But, I wanted also to bounce my own songs and to play over so, anyway, it's a basic tool to grow as a guitarist, without having a real band with which to share experiences.
I am using an MP3 player (Sony Walkman) to store and reproduce the backing tracks and, until very recently, I was using some Hi-Fi set to reproduce the songs of the Walkman thru its auxiliary input.
The basic issue with Hi-Fi chains is their lack of power, if we compare them to the power of any tubular amp. Even with 150W by channel, I cannot get a minimum volume to deal with the weakest of my amps.
Probably, the real issue are the HiFi Speakers, clearly less efficient that the speakers mounted in guitar amps.
For a while, I went to some rental rooms and, they had there some cheap PA systems that, even that they weren't powerful enough, they delivered way more power than the Hi-Fi sets I've tested. Probably, because PA speakers are bigger and more efficient.
So, I wanted to go for an affordable solution that helped me with backing tracks.
I've browsed several solutions. By example, Peavey was a good candidate and, it's an excellent guitar amps builder. Yamaha always have "correct" products, with quality enough, maybe not the cream of the cream but they barely disappoint me. They, they were cheaper offers, as the T.Racks (white brand from Thomann) or Behringer, that were offering more power for less price.
But, my own experience with Behringer's PA systems made me to run away that idea. The T.Racks took my attention but, they were way over or lower the power that I was looking for (in fact, my target power wasn't the right one).
Peavey was short in power and, Yamaha seemed to have the right power I was after and, I usually rely on their products (everything I've tested from Yamaha satisfied my expectations, taking into account it price) so, I finally went for a Yamaha StagePass 300 PA System.
I had to buy also a couple of stands for those speakers and a couple of StagePass adaptors to mount the Speakers on those stands.
Presentation
Everything comes well packet in a huge box, well protected against shocks.
One of the thing that more confused me is that, when I've opened the box I've saw just two Speakers... I was wondering where the amplifier was!.
Both Speakers have the rear side carved, leaving an useful box that is finished with a tap.
First Speaker has the amplifier inserted in its cavity (which you can easily remove) and, the second speaker has room for speaker's cables and even for some mic.
So, despite of the stands, you can easily transport the whole PA system with just two hands, one for each speaker. Very nice idea, indeed.
The upper handles of speakers make their transport really easy so, it's very comfortable to move "all the gear" and, it's quick to deploy and mount everything.
The Kit includes a couple of speaker cables, with mono jacks, of about 3 meters long. This is a bit short for a PA System so, take this into account if you need longer cables. And, watch out, you can use just mono jack speaker cables, there is just a mono jack plug in each speaker.
The amp-mixer has 4 mono channels and 2 stereo channels. Mono channels admit mic or line/instrument inputs (switchable) and, you can send each channel to a global reverberation effect, which level is common to all channels.
Every channel has a 2-bands equalizer (treble and Bass).
It has outputs for passive speakers (as the ones included in the kit), for active speakers and, a line output.
The output level for passive speakers is being controlled with the Master knob, while the other two outputs are being regulated with the Monitor knob.
It has a switch named speech/music, that allows to change the operating mode of the amp to make it more efficient for speeching or to play music.
It's made in China (as practically everything, nowadays), has an excellent finish and, seems to be built like a tank, with quality components.
How it sounds
No doubt in this department: it sounds EXCELLENTLY!. All frequencies are well represented and, even that theoretically should have weak basses (according to the nominal frequencies covered as per technical specifications), I think it has a spectacular kick in basses.
Compared to how those Behringer sound, no discussion. The Berhinger I've tested sound boxy and blurred, while the StagePass 300 is clearly representing every frequency (even those that theoretically cannot).
Powerful enough?
Well, here we are again. It could seem that a PA system having 150W per channel should be a sonic bomb, taking into account that the powerfuller of my amps has "just" 50W but, as ever, Watts doesn't tell us the whole history. What at the end matters is the efficiency of speakers.
Even that these speakers can deliver 112dB@1 m, it's clear that the amp-mixer cannot feed them with meat enough.
As most of amp-mixers, it has a display with leds that correspond to the output level and, sure, as any solid state device, to constantly maintain the output level in the "red zone" ends frying the amp. So, you have to limit the output to safer (therefore, quieter) output levels.
The Volume knobs of channels and Master, have a mark around the 75% of their dials, what sets the limit of the safer area (distortion free and without overloading the amp circuitry).
Probably, the amp will sound its best at those levels, without saturation neither inner clipping but, at those levels this PA cannot sound louder han a HI-FI mini-chain. It moves more air and has more basses, that's all.
Real thing is that I've forced the channel level to its max and, leaved the Master just a hair over the "mark", controlling that the clipping indicator isn't permanently on. This provides me an acceptable loudness.
If I krank even the little Vox Night Train, the tube amp totally covers the sound of the PA. To be able to use the PA System, I have to play every amp at low levels. Therefore, the help of some pushing pedals is a must in this environment.
Conclusion
I honestly don't think this particular PA System can cover the levels required in a band situation. I don't see this gear able to compete against a bass guitar, a drum kit or a guitar. It can be useful for small conferences, for very small groups of people and, small rooms.
It's excellent to have a party with your friends and, it's good to play at home, also, every time that you can tame the loudness of your tube amp or that you can play with a solid state amp. But, if you like to push your tubes amp, this System cannot compete with your amp.
Related to quality of finished and sound, it's excellent for the price and, I don't regret having purchased it.
Related to transportability, excellent solution. Very well designed, comfortable and with nice-look.
Related to power, a bit limited and, quite well close to a HI-FI chain
Clearly, 150 Solid State Watts are VERY FEW watts to compete against 15 tubular Watts. I'm suspecting that I would need from 300 to 500 W per channel to be able to krank my amps to an acceptable loudness, to reach their sweet spots.
Anyway, I had an small gig, with very few people in a small room and, the power of the StagePass was enough to play with the Vox Night Train, with an awesome quality in sound.
Note: this entry was already published in my old Spanish blog, during February/2012. I am just revisiting it with added info, based on the experience more than one year later.
To practice with backing tracks is interesting to me for several reasons. In first place, I avoid to go boring playing the guitar alone without any particular target. In second place, this helps me to "couple" my sound and rig to a "virtual band of musics".
But, I wanted also to bounce my own songs and to play over so, anyway, it's a basic tool to grow as a guitarist, without having a real band with which to share experiences.
I am using an MP3 player (Sony Walkman) to store and reproduce the backing tracks and, until very recently, I was using some Hi-Fi set to reproduce the songs of the Walkman thru its auxiliary input.
The basic issue with Hi-Fi chains is their lack of power, if we compare them to the power of any tubular amp. Even with 150W by channel, I cannot get a minimum volume to deal with the weakest of my amps.
Probably, the real issue are the HiFi Speakers, clearly less efficient that the speakers mounted in guitar amps.
For a while, I went to some rental rooms and, they had there some cheap PA systems that, even that they weren't powerful enough, they delivered way more power than the Hi-Fi sets I've tested. Probably, because PA speakers are bigger and more efficient.
So, I wanted to go for an affordable solution that helped me with backing tracks.
I've browsed several solutions. By example, Peavey was a good candidate and, it's an excellent guitar amps builder. Yamaha always have "correct" products, with quality enough, maybe not the cream of the cream but they barely disappoint me. They, they were cheaper offers, as the T.Racks (white brand from Thomann) or Behringer, that were offering more power for less price.
But, my own experience with Behringer's PA systems made me to run away that idea. The T.Racks took my attention but, they were way over or lower the power that I was looking for (in fact, my target power wasn't the right one).
Peavey was short in power and, Yamaha seemed to have the right power I was after and, I usually rely on their products (everything I've tested from Yamaha satisfied my expectations, taking into account it price) so, I finally went for a Yamaha StagePass 300 PA System.
I had to buy also a couple of stands for those speakers and a couple of StagePass adaptors to mount the Speakers on those stands.
Presentation
Everything comes well packet in a huge box, well protected against shocks.
One of the thing that more confused me is that, when I've opened the box I've saw just two Speakers... I was wondering where the amplifier was!.
Both Speakers have the rear side carved, leaving an useful box that is finished with a tap.
First Speaker has the amplifier inserted in its cavity (which you can easily remove) and, the second speaker has room for speaker's cables and even for some mic.
So, despite of the stands, you can easily transport the whole PA system with just two hands, one for each speaker. Very nice idea, indeed.
The upper handles of speakers make their transport really easy so, it's very comfortable to move "all the gear" and, it's quick to deploy and mount everything.
The Kit includes a couple of speaker cables, with mono jacks, of about 3 meters long. This is a bit short for a PA System so, take this into account if you need longer cables. And, watch out, you can use just mono jack speaker cables, there is just a mono jack plug in each speaker.
The amp-mixer has 4 mono channels and 2 stereo channels. Mono channels admit mic or line/instrument inputs (switchable) and, you can send each channel to a global reverberation effect, which level is common to all channels.
Every channel has a 2-bands equalizer (treble and Bass).
It has outputs for passive speakers (as the ones included in the kit), for active speakers and, a line output.
The output level for passive speakers is being controlled with the Master knob, while the other two outputs are being regulated with the Monitor knob.
It has a switch named speech/music, that allows to change the operating mode of the amp to make it more efficient for speeching or to play music.
It's made in China (as practically everything, nowadays), has an excellent finish and, seems to be built like a tank, with quality components.
How it sounds
No doubt in this department: it sounds EXCELLENTLY!. All frequencies are well represented and, even that theoretically should have weak basses (according to the nominal frequencies covered as per technical specifications), I think it has a spectacular kick in basses.
Compared to how those Behringer sound, no discussion. The Berhinger I've tested sound boxy and blurred, while the StagePass 300 is clearly representing every frequency (even those that theoretically cannot).
Powerful enough?
Well, here we are again. It could seem that a PA system having 150W per channel should be a sonic bomb, taking into account that the powerfuller of my amps has "just" 50W but, as ever, Watts doesn't tell us the whole history. What at the end matters is the efficiency of speakers.
Even that these speakers can deliver 112dB@1 m, it's clear that the amp-mixer cannot feed them with meat enough.
As most of amp-mixers, it has a display with leds that correspond to the output level and, sure, as any solid state device, to constantly maintain the output level in the "red zone" ends frying the amp. So, you have to limit the output to safer (therefore, quieter) output levels.
The Volume knobs of channels and Master, have a mark around the 75% of their dials, what sets the limit of the safer area (distortion free and without overloading the amp circuitry).
Probably, the amp will sound its best at those levels, without saturation neither inner clipping but, at those levels this PA cannot sound louder han a HI-FI mini-chain. It moves more air and has more basses, that's all.
Real thing is that I've forced the channel level to its max and, leaved the Master just a hair over the "mark", controlling that the clipping indicator isn't permanently on. This provides me an acceptable loudness.
If I krank even the little Vox Night Train, the tube amp totally covers the sound of the PA. To be able to use the PA System, I have to play every amp at low levels. Therefore, the help of some pushing pedals is a must in this environment.
Conclusion
I honestly don't think this particular PA System can cover the levels required in a band situation. I don't see this gear able to compete against a bass guitar, a drum kit or a guitar. It can be useful for small conferences, for very small groups of people and, small rooms.
It's excellent to have a party with your friends and, it's good to play at home, also, every time that you can tame the loudness of your tube amp or that you can play with a solid state amp. But, if you like to push your tubes amp, this System cannot compete with your amp.
Related to quality of finished and sound, it's excellent for the price and, I don't regret having purchased it.
Related to transportability, excellent solution. Very well designed, comfortable and with nice-look.
Related to power, a bit limited and, quite well close to a HI-FI chain
Clearly, 150 Solid State Watts are VERY FEW watts to compete against 15 tubular Watts. I'm suspecting that I would need from 300 to 500 W per channel to be able to krank my amps to an acceptable loudness, to reach their sweet spots.
Anyway, I had an small gig, with very few people in a small room and, the power of the StagePass was enough to play with the Vox Night Train, with an awesome quality in sound.
13 April 2013
Home Studio: Primacoustics London 12A Studio Kit: take the control of your room?
Introduction
Note: this entry was already published in my old Spanish version of this blog, around February/2011. I'm just revisiting it here.
Well, I am still trying to correct the acoustics of my mixing room. I was really scared seen as any mix that sounds ok thru my near field monitors, sounds really bad when bounced to an MP3 file.
The IK Multimedia ARC system helps a bit but, it delivers a "washed" sound and, even that it helps to correct some deficiencies of the room, it cannot get rid of issues derived of excessive early reflexions (fluttering echo, comb filter...) and, cannot work for low frequencies with the same efficiency that a real Bass Trap.
In my recording tests with vocals (and the new Rode NT2-A, that makes everything to pop up in the track), I've notices that some reflexions even doubled some words, with a temporal shift really ugly, making the resulting take a bit blurred. Even that not all the track was affected, there are some parts were the excessive early reflexions created real issues.
The T.Bone Micscreen filter seems to help nothing to protect the mic from early reflexions. The mic is highly sensible and gets everything with ease.
In this situation, I was forced to go an step further and to try to acoustically treat my room; something that I was avoiding because of two fundamental reasons: the high cost and the "aesthetic modification".
But... how to start, when you have not enough technical knowledge?.
After some time reading articles and reviewing prices, it seemed to me an acceptable idea to try to get rid of those problems by purchasing some of those kits for little home studios.
Clearly, no specialist will recommend you to go this way. Most those that "support" the idea, will coincide on that a solution based on high density foam pieces DOESN'T solves anything and, has a durability issue (it seems that the Sun light among other environmental agents attack the foam, which ends getting rid, with a ugly impact in your Studio look).
As a minor evil, they will "accept" solutions not based in high density foam.
So, after browsing several offerings of that kind, it seemed to me that the best option was the Primacoustic London 12A Studio Kit. In first place, the panels were made of glass fiber (instead of foam) and, in a second place, it seemed to come with an anchoring system, what allows you to remove the panels and reuse them in any other room. Since I'm in a rental house, to me is a must to have a solution that can be removable and reusable in any other house.
With a real crack in the credit of my card, I bought the kit and, this is my experience.
Presentation
The whole kit comes packet in a single box of huge dimensions, which size comes determined by the size of the "bass control panels". It comes well packed and includes 100 studs and 100 screws, to fix the anchoring pieces to the wall and, also a drill (curious!). Finally, all the needed anchoring pieces come included in the pack.
A nice triptych, with lot of propaganda about their wonderful products and, no user's manual or instructions that help you to guess "how to do it". Luckily, they have a guide of installation in their website, very clear and easy to follow, among some demonstrative video and, therefore, the plannification of the installation is clear.
Panels
Well, the first thing I wanted to snoop were panels... what are they made about? how are they built? what do they have in special?.
First disappointment. Panels are made from compacted glass fiber wool, coated with some kind of clear substance (epoxy, I guess), to avoid the wool to frying. That panel is then wrapped in a fabric piece (as per their ad, acoustically neutral), which is glued to the back side of the panel.
A curious thing, the product says that the panels were made in China, while the assembling was made in Canada. What do the Canadian people so good that Chinese people cannot do in something so simple?. Is that just a way to justify the high price of a cheap product?.
Well, the square meter of glass fiber in a Depot, costs about 2 Eur!!!.
The 8 "control columns" make about 3 square meters.
The 2 "bass control columns" make about 1,5 square meters.
The 12 "tiles" make about 1,1 square meters.
So, summing it everything up, at 2 Eur / square meter, we are talking about 11,2 Eur !!!.
Ok, then we have the steel anchoring pieces. About 28 of "regular" ones and 8 "special" for the squared panels. Primacoustic sells 24 broadway impalers (that's how they call the anchoring pieces) at 46 Eur. So, a couple of pack make 92 Eur.
The 8 squaring impalers are being sold at 69 Eur.
Summing everything together, with the fiber panels, it means 172,2 Eur.
Let add some since fabric, at 5 Eur / meter and, let say that we will need 3 times the linear meters of all the panels, about 18 meters, at 5 Eur, means 90 Eur.
Ok, that makes 262,2 Eur in material, at the price that is being sold in stores.
So, summing up everything together, with the epoxy rosin, the glue to glue the fabric to the panel, etc., for around 300 Eur, if you have good hands, you can do it by yourself and, you can even put there the fabric that more likes to you because, at the end, those panels can decorate or break the look of your room.
If they were made totally in China, since the price of the human work is negligible and, they send containers of material (lowering the shipping costs astronomically), maybe we had a finished product for about 400 Eur.
Up to the 722 that actually costs... help me to understand the difference?.
To be honest, the look of panels is good, very classy but, they smell really bad. For this price, they could add some pleasant Asian essence.
Installation
Well, I have to recognize that, even that it takes its time, it's easy but... is there any need for so much complexity?.
I've weighted panels on hand and, even the huger are lightweight, easy to handle with a single hand.
The anchoring system consists into put one impaler for each small tile, 2 impalers for "control columns" and 4 special impalers for the squared "bass control columns".
The "regular" impalers need 2 screws, the "specials" 4 (but, I've installed those impalers with 2 in diagonal).
Screws and studs are of size 6, what sincerely is unnecessary for the weight of those panels.
If you sum up, 8 columns need 2 x 2 x 8 = 32 screws, 2 bass columns need 2 x 4 x 4 = 16 screws and, the 12 tiles need 12 x 2 = 24 screws. 72 screws in total, 72 holes in your walls!!!.
IMHO, every panels, except maybe the bass control columns, could be placed on the wall by using an industrial Velcro solution, which will reduce the installation time dramatically. The only panels that probably still need the impalers are those squared bass control columns, because of its very special position.
But, if you want to go with the recommended installation way, then be patient.
I've thrown 3 straight lines by using 6 pushpins and a thin rope, that gave the upper adjust level for pannels, the placement for the upper impaler and the position for the lower impaler.
And, once mounted, what it looks like?
Honestly, it gives to the room a very "professional" look. Panels are well finished and, the chosen fabric has a very neutral color that suits very will to the room.
From Theory to Practice
LEDE Studio?
The two basis ideas that "justify" this kit are: stereo balance and the creation of a LEDE room.
What the heck is a LEDE room?
LEDE means Live End / Dead End.
The concept consists into split the room in two very differentiated areas of same dimensions, a live zone (with full reflexions) and a dead zone (with no reflexions).
The most of the kit should be installed in the dead zone (around the hearing spot), to "kill" the early reflexions, that are the bad boys of this tale.
In the living zone, we will install just the little tiles to aid sound diffusion.
All this is very nice, indeed but, everybody has it's own limitations respect of its own Home Studio, usually placed in the less used room and, with different furniture, windows, etc.
So, In my case, I just did what my room allowed me. Even that I understand the theory behind, I've preferred to "put all the meat" in the side walls and back wall of the hearing spot, since neither the front window, the heaters or the two doors leave me space for other kind of arrangement.
So, I recognize that my room isn't so LEDE but, my woman said that it looks impressive.
So good, so nice but... does it works?
After the hard work made with installation, it was too like to test my "new" studio. This and that I was afraid to be disappointed with results, which made me nervous thinking in the amount of credit that I had to restore spent in crap.
Well, it took me one more day to start my tests. I've switched on the audio card, monitors and, loaded the DAW with the mix I was working on before starting all this adventure.
Er?... great part of the "boom-boom" disappeared and, it seems easier to work with reverberations now. I mix with near field monitors and, do a bounce of the mix to an MP3 file.
Nice!. For very first time, the sound of the MP3 is closer to the sound that I've heard thru monitors.
The Mix have some issues still but, this time the mids doesn't seem so inflated and lows doesn't produce me a headache.
Placebo effect?.
My analytic brain gets the control of my heart and, the devil in my left ear says to me: "hey, maybe you WANT to hear some enhancement just because all these costed a kidney. Get that ARC mic and check if equalization issues were corrected!".
So... one more day to start testing with the help of the ARC system.
I mount the mic, start the application and, patiently I am taking the 24 samples around the hearing spot.
And, here we are the two comparison pictures.
Firstly, how everything was before trying to save the world with that Primacoustic Kit:
And, now, the picture after the "treatment":
The Orange lines correspond to the deviation of the room sound respect to the target curve (plane eq, Green line). White lines correspond to the correction curve that ARC can achieve.
In my understanding, already existing anomalies are still there and, they even increased.
The only part that seems to be enhanced is the "AIR" band, that seems to be recovered about 6 dB.
It seems also that the zone of mid-trebles, were the second order harmonics and presence is being represented is now bumped up, while they were already OK BEFORE (plain response).
The response in low frequencies seems to had been dimmed, also.
I've got very mixed sensations and thoughts. In one side, it's true that I feel a clear difference to best. In the room the low frequencies were really boomy and, I feel this was fixed in around an 80%. The enhancement is clearly audible, whatever the ARC graphic says. If that enhancement is due to this recovering of the AIR band, I dunno.
Where I am hearing the best improvement is related to reflexions. Clearly, this is working well against excessive reverberations.
Sincerely, NOT, I DON'T. Not to this one and no other. You are paying a lot of money for very few enhancement. This kit can help with reverberations and, maybe, it's increasing the AIR band but, isn't able to get rid of the room modes, that are one of the biggest issues to have a good mix.
Since it is clear that with material of high density you can reduce the early reflexions, there are many other ways to achieve this effect at a lower cost and, you can build them by yourself: glass fiber panels, dense curtains, dense carpets... anything very massive, with a very irregular surface that can increase the 3D face of the wall can help to reduce early reflexions.
However, to control the modal issues of your room, it seems that the only medicine are Bass Trap. Bob
Note: this entry was already published in my old Spanish version of this blog, around February/2011. I'm just revisiting it here.
Well, I am still trying to correct the acoustics of my mixing room. I was really scared seen as any mix that sounds ok thru my near field monitors, sounds really bad when bounced to an MP3 file.
The IK Multimedia ARC system helps a bit but, it delivers a "washed" sound and, even that it helps to correct some deficiencies of the room, it cannot get rid of issues derived of excessive early reflexions (fluttering echo, comb filter...) and, cannot work for low frequencies with the same efficiency that a real Bass Trap.
In my recording tests with vocals (and the new Rode NT2-A, that makes everything to pop up in the track), I've notices that some reflexions even doubled some words, with a temporal shift really ugly, making the resulting take a bit blurred. Even that not all the track was affected, there are some parts were the excessive early reflexions created real issues.
The T.Bone Micscreen filter seems to help nothing to protect the mic from early reflexions. The mic is highly sensible and gets everything with ease.
In this situation, I was forced to go an step further and to try to acoustically treat my room; something that I was avoiding because of two fundamental reasons: the high cost and the "aesthetic modification".
But... how to start, when you have not enough technical knowledge?.
After some time reading articles and reviewing prices, it seemed to me an acceptable idea to try to get rid of those problems by purchasing some of those kits for little home studios.
Clearly, no specialist will recommend you to go this way. Most those that "support" the idea, will coincide on that a solution based on high density foam pieces DOESN'T solves anything and, has a durability issue (it seems that the Sun light among other environmental agents attack the foam, which ends getting rid, with a ugly impact in your Studio look).
As a minor evil, they will "accept" solutions not based in high density foam.
So, after browsing several offerings of that kind, it seemed to me that the best option was the Primacoustic London 12A Studio Kit. In first place, the panels were made of glass fiber (instead of foam) and, in a second place, it seemed to come with an anchoring system, what allows you to remove the panels and reuse them in any other room. Since I'm in a rental house, to me is a must to have a solution that can be removable and reusable in any other house.
With a real crack in the credit of my card, I bought the kit and, this is my experience.
Presentation
The whole kit comes packet in a single box of huge dimensions, which size comes determined by the size of the "bass control panels". It comes well packed and includes 100 studs and 100 screws, to fix the anchoring pieces to the wall and, also a drill (curious!). Finally, all the needed anchoring pieces come included in the pack.
A nice triptych, with lot of propaganda about their wonderful products and, no user's manual or instructions that help you to guess "how to do it". Luckily, they have a guide of installation in their website, very clear and easy to follow, among some demonstrative video and, therefore, the plannification of the installation is clear.
Panels
Well, the first thing I wanted to snoop were panels... what are they made about? how are they built? what do they have in special?.
First disappointment. Panels are made from compacted glass fiber wool, coated with some kind of clear substance (epoxy, I guess), to avoid the wool to frying. That panel is then wrapped in a fabric piece (as per their ad, acoustically neutral), which is glued to the back side of the panel.
A curious thing, the product says that the panels were made in China, while the assembling was made in Canada. What do the Canadian people so good that Chinese people cannot do in something so simple?. Is that just a way to justify the high price of a cheap product?.
Well, the square meter of glass fiber in a Depot, costs about 2 Eur!!!.
The 8 "control columns" make about 3 square meters.
The 2 "bass control columns" make about 1,5 square meters.
The 12 "tiles" make about 1,1 square meters.
So, summing it everything up, at 2 Eur / square meter, we are talking about 11,2 Eur !!!.
Ok, then we have the steel anchoring pieces. About 28 of "regular" ones and 8 "special" for the squared panels. Primacoustic sells 24 broadway impalers (that's how they call the anchoring pieces) at 46 Eur. So, a couple of pack make 92 Eur.
The 8 squaring impalers are being sold at 69 Eur.
Summing everything together, with the fiber panels, it means 172,2 Eur.
Let add some since fabric, at 5 Eur / meter and, let say that we will need 3 times the linear meters of all the panels, about 18 meters, at 5 Eur, means 90 Eur.
Ok, that makes 262,2 Eur in material, at the price that is being sold in stores.
So, summing up everything together, with the epoxy rosin, the glue to glue the fabric to the panel, etc., for around 300 Eur, if you have good hands, you can do it by yourself and, you can even put there the fabric that more likes to you because, at the end, those panels can decorate or break the look of your room.
If they were made totally in China, since the price of the human work is negligible and, they send containers of material (lowering the shipping costs astronomically), maybe we had a finished product for about 400 Eur.
Up to the 722 that actually costs... help me to understand the difference?.
To be honest, the look of panels is good, very classy but, they smell really bad. For this price, they could add some pleasant Asian essence.
Installation
Well, I have to recognize that, even that it takes its time, it's easy but... is there any need for so much complexity?.
I've weighted panels on hand and, even the huger are lightweight, easy to handle with a single hand.
The anchoring system consists into put one impaler for each small tile, 2 impalers for "control columns" and 4 special impalers for the squared "bass control columns".
The "regular" impalers need 2 screws, the "specials" 4 (but, I've installed those impalers with 2 in diagonal).
Screws and studs are of size 6, what sincerely is unnecessary for the weight of those panels.
If you sum up, 8 columns need 2 x 2 x 8 = 32 screws, 2 bass columns need 2 x 4 x 4 = 16 screws and, the 12 tiles need 12 x 2 = 24 screws. 72 screws in total, 72 holes in your walls!!!.
IMHO, every panels, except maybe the bass control columns, could be placed on the wall by using an industrial Velcro solution, which will reduce the installation time dramatically. The only panels that probably still need the impalers are those squared bass control columns, because of its very special position.
But, if you want to go with the recommended installation way, then be patient.
I've thrown 3 straight lines by using 6 pushpins and a thin rope, that gave the upper adjust level for pannels, the placement for the upper impaler and the position for the lower impaler.
And, once mounted, what it looks like?
Honestly, it gives to the room a very "professional" look. Panels are well finished and, the chosen fabric has a very neutral color that suits very will to the room.
From Theory to Practice
LEDE Studio?
The two basis ideas that "justify" this kit are: stereo balance and the creation of a LEDE room.
What the heck is a LEDE room?
LEDE means Live End / Dead End.
The concept consists into split the room in two very differentiated areas of same dimensions, a live zone (with full reflexions) and a dead zone (with no reflexions).
The most of the kit should be installed in the dead zone (around the hearing spot), to "kill" the early reflexions, that are the bad boys of this tale.
In the living zone, we will install just the little tiles to aid sound diffusion.
All this is very nice, indeed but, everybody has it's own limitations respect of its own Home Studio, usually placed in the less used room and, with different furniture, windows, etc.
So, In my case, I just did what my room allowed me. Even that I understand the theory behind, I've preferred to "put all the meat" in the side walls and back wall of the hearing spot, since neither the front window, the heaters or the two doors leave me space for other kind of arrangement.
So, I recognize that my room isn't so LEDE but, my woman said that it looks impressive.
So good, so nice but... does it works?
After the hard work made with installation, it was too like to test my "new" studio. This and that I was afraid to be disappointed with results, which made me nervous thinking in the amount of credit that I had to restore spent in crap.
Well, it took me one more day to start my tests. I've switched on the audio card, monitors and, loaded the DAW with the mix I was working on before starting all this adventure.
Er?... great part of the "boom-boom" disappeared and, it seems easier to work with reverberations now. I mix with near field monitors and, do a bounce of the mix to an MP3 file.
Nice!. For very first time, the sound of the MP3 is closer to the sound that I've heard thru monitors.
The Mix have some issues still but, this time the mids doesn't seem so inflated and lows doesn't produce me a headache.
Placebo effect?.
My analytic brain gets the control of my heart and, the devil in my left ear says to me: "hey, maybe you WANT to hear some enhancement just because all these costed a kidney. Get that ARC mic and check if equalization issues were corrected!".
So... one more day to start testing with the help of the ARC system.
I mount the mic, start the application and, patiently I am taking the 24 samples around the hearing spot.
And, here we are the two comparison pictures.
Firstly, how everything was before trying to save the world with that Primacoustic Kit:
And, now, the picture after the "treatment":
Comparing both pictures, I don't know if to wear the hat with donkey ears or to hara-kiri myself.
In the Right side of the room, we can see an increase in the mid-low frequencies, respect of the original situation and, this is because I've CENTERED the desk in the room, looking for that nice stereo balance and, now I have very similar issues (very balanced, indeed) in both sides.
More about it, the already existing peaks in the region between 100 - 200 Hz seems to had been increased.
In the Right side of the room, we can see an increase in the mid-low frequencies, respect of the original situation and, this is because I've CENTERED the desk in the room, looking for that nice stereo balance and, now I have very similar issues (very balanced, indeed) in both sides.
More about it, the already existing peaks in the region between 100 - 200 Hz seems to had been increased.
The Orange lines correspond to the deviation of the room sound respect to the target curve (plane eq, Green line). White lines correspond to the correction curve that ARC can achieve.
In my understanding, already existing anomalies are still there and, they even increased.
The only part that seems to be enhanced is the "AIR" band, that seems to be recovered about 6 dB.
It seems also that the zone of mid-trebles, were the second order harmonics and presence is being represented is now bumped up, while they were already OK BEFORE (plain response).
The response in low frequencies seems to had been dimmed, also.
Conclusions
I've got very mixed sensations and thoughts. In one side, it's true that I feel a clear difference to best. In the room the low frequencies were really boomy and, I feel this was fixed in around an 80%. The enhancement is clearly audible, whatever the ARC graphic says. If that enhancement is due to this recovering of the AIR band, I dunno.
Where I am hearing the best improvement is related to reflexions. Clearly, this is working well against excessive reverberations.
Does this kit solves the modal issues?
NOT, NOT and RE-NOT.
Every small room suffers of some resonance modes that accentuate frequencies between 120 - 200 Hz (more or less) and, it's clear that the standard control panels aren't able to get rid of this messing issue.
Those modes make the sound very confusing, with bad defined lows, boomy, very ear fatiguing.
According to every expert, only Real Bass Traps can solve this issue.
Those modes make the sound very confusing, with bad defined lows, boomy, very ear fatiguing.
According to every expert, only Real Bass Traps can solve this issue.
But, the price of a single Bass Trap is scary, above 400 Eur and, at least, you need two of them.
Do you recommend me this kit?
Sincerely, NOT, I DON'T. Not to this one and no other. You are paying a lot of money for very few enhancement. This kit can help with reverberations and, maybe, it's increasing the AIR band but, isn't able to get rid of the room modes, that are one of the biggest issues to have a good mix.
Since it is clear that with material of high density you can reduce the early reflexions, there are many other ways to achieve this effect at a lower cost and, you can build them by yourself: glass fiber panels, dense curtains, dense carpets... anything very massive, with a very irregular surface that can increase the 3D face of the wall can help to reduce early reflexions.
However, to control the modal issues of your room, it seems that the only medicine are Bass Trap. Bob
Katz (among others) recommend Real Traps by Mondo.
In my honest opinion, spent that money in good Bass Traps and, reduce rest of issues with cheaper and imaginative solutions.
One more thing to take into account.
I wrote to Primacoustic people with an sketch of my room, including furniture and, rest of elements, together with the resulting curves of measures before and after and, asking them for some answers about results and, asking them for some guidance about how to get better results with their kit.
I WAS ABSOLUTELLY IGNORED.
WHAT A SHIT OF CUSTOMER' SERVICE HAS PRIMACOUSTICS.
12 April 2013
Home Studio: Rode NT2-A Studio Solution Set and T.Bone Mic Screen
Introduction
Note: this entry was already published in my old Spanish version of this blog, around January/2011. I am revisiting it here.
One of the most complex things for a Home Studio is to get a good record of clean and defined vocals.
The typical mic to start, that anyone will recommend you is the Shure SM58.
Honestly, I didn't liked any of the takes that I did with this mic. To me, vocals sound so close and dark and, the track needs to be hardly re-EQ'd to make it to sound similar to the real thing.
Overall, the mic lacks some gain, what forces you to sing closer to the mic, increasing the proximity effect of such a mic.
During my visits to a friend' studio, I had the opportunity to try a mic Rode (probably, one NT2000, memory fails here). That mic impressed me because of its clarity and detail so, I wanted to substitute my SM58 with some model from Rode.
I bought the NT2-A because of economical reasons. As per the articles of Paul White (SOS), there are just two levels of large diaphragm mics: those that cost a kidney (High end Neuman, AKG...) and the "affordable" ones.
The gap between "affordable" and the mythical Neuman U87 is really huge but, differences in price between the "economics" are less dramatic. After reading an article around the NT2-A, I've been convinced that this was the best solution for my financial state.
Paul White says that this is the mic with less floor noise of the affordable ones, valid for practically anything and, with a very plain response in frequencies. A bright mic that can help to opaque voices, as mine... so... the dices were rolling.
In other side, it's clear that there is a clear difference between a dynamic mic, as the Shure SM57 or SM58, that are designed to catch just the sound very close to its capsule, highly attenuating the sound with the distance and, a large diaphragm mic, as the Rode NT2-A, designed to catch everything around.
Trying to reduce the ambience noises that this mic can catch, I've tried also an absorbent mic screen, concretely, the T.Bone Micscreen.
Follows, my impressions about both things.
Rode NT2-A Studio Solution Set
This nice pack includes the mic NT2-A, the anti-shock mount, an integrated pop filter, an XLR cable, a CD with more marketing than useful info and, a fund for the mic.
First pack that I've received was faulty. The Omni and Eight mode of the mic were working ok related to gain but, they had a lot of floor noise produced by the mic itself, that was far away from that quietness that Paul White was talking about.
The cardiod mode didn't work and the floor noise was really high, with a very weak signal.
So, I did my first tests with the Eight mode and, despite of the floor noise, I was really impressed about how clear and defined were vocals and, best of all, no need to re-EQ the track (except a small dip around 5KHz, to remove some piercing presence).
After a while, I had the replacement pack, with a new mic and... this time was everything working fine!.
Impressive the low level of the floor noise and, impressive the high sensibility of this mic, able to catch the quietest sound happening in the room, even out of the cardioid pattern.
Also, impressive gain level. If my tests with the first faulty mic needed to raise the input gain in the pre-amp, for those tests with the second one, I needed to maintain the gain knob below noon, to avoid the bus to clip.
It's able to suffer high level of sound pressure, accepting 137 dB (normal mode) or, up to 147 dB (with pad set to -10dB). I've started to scream like a monkey asking for an autograph to Cheetah and, the mic get it all, without any distortion. Cool.
Fortunately, the pack include the antishock-mount, since this mic needs to be mount in a ring and, otherwise, I hadn't the opportunity to try it mounted in my mic stand.
The anti-pop filter is just correct and, a bit limited in movements. Possibly, better to buy an anti-pop filter with a flexible arm.
The antishock-mount seems reliable and strong.
The mic weight considerably and has an impressive size. Done to dynamic mics, as the SM57, SM58 and, some Sennheiser, to stand this mic on my hands was a religious experience.
Overall, it looks really professional. It doesn't seems a cheap gear produced at low cost in China and, in fact, Rode' series ending in A are being produced in Australia, where Rode were able to reduce costs by using sophisticated machines that reduced the mounting time and, excess of personal.
The takes I did with this mic (even the faulty one), leaved a track with a nice gain, awesome dynamic range, lot of detail and defined vocals that suited the mix naturally. I had to dip around 5K to remove some excessive bright there but, even without touching the EQ, the result is directly usable.
I am really happy with this mic, indeed.
Due to its high ability to get high preaure level, it's even possible to use it to record guitar or bass amp, and even Kick drums. To do that, we need to switch on the option -10dB pad.
So, a great mic, with an affordable price related to other large diaphragm mics, high quality, impecable finishing, very low floor noise, plain EQ and, the possibilty to use it for any micking task in the Studio.
To whoever that is sick about its SM57 or SM58, I highly recommend him to check this mic!.
T.Bone MicScreen
Well, I've tried to filter part of the ambience sound heard in the back side of the mic (not cardioid zone) and, to avoid room reflexions, by using this mic screen, because it was really affordable and, opinions were very mixed about this kind of solutions.
Comes unmounted but, the task is easy. Everything comes in a box with a very professional look. Has some interesting accessories, as a bar that allows you to mount two mics (for stereo tasks).
The screen as an adjustable wide (but, unstable and not accurate) and, it's easy to couple to any mic stand but, because of its weight, you better choose a stand with a weighty base.
Related to its function... does it really work absorbing the ambience noise?.
Answer is... NOT!.
I've put the stand with the mic facing a books shell (to avoid reflexions) and, with the screen behind, "protecting" the rear side of the mic (not cardioid).
In first place, the NT2-A is cardioid but not hypercardioid.
I've tried to click my fingers in the back zone and, it catches the sound with total clarity, with a volume level slightly lower respect to same sound in front the mic but, high enough.
Then, I've tried the same behind the screen and, the mic was catching the sound without any issues.
Vocals tests were also very clear about all this.
While I was adjusting the gain level, in my PC, far away from the mic and, with headphones on, I was able to hear the sound that I was producing on my desk, while working.
Therefore, in my honest opinion, probably can help to reduce the reflexions that catch the mic on its rear side but, in no way its absorbing the ambiental noise.
In the track, I was able to clearly hear the reflexions in the room (not acoustically treated).
The good thing is that, mounted on the stand and, with that big NT2-A, you seem like an astronaut handling a sophisticated device for spatial exploration and, this can help you to pick up girls. Take a picture of yourself with all this gear and then... forget the screen.
Note: this entry was already published in my old Spanish version of this blog, around January/2011. I am revisiting it here.
One of the most complex things for a Home Studio is to get a good record of clean and defined vocals.
The typical mic to start, that anyone will recommend you is the Shure SM58.
Honestly, I didn't liked any of the takes that I did with this mic. To me, vocals sound so close and dark and, the track needs to be hardly re-EQ'd to make it to sound similar to the real thing.
Overall, the mic lacks some gain, what forces you to sing closer to the mic, increasing the proximity effect of such a mic.
During my visits to a friend' studio, I had the opportunity to try a mic Rode (probably, one NT2000, memory fails here). That mic impressed me because of its clarity and detail so, I wanted to substitute my SM58 with some model from Rode.
I bought the NT2-A because of economical reasons. As per the articles of Paul White (SOS), there are just two levels of large diaphragm mics: those that cost a kidney (High end Neuman, AKG...) and the "affordable" ones.
The gap between "affordable" and the mythical Neuman U87 is really huge but, differences in price between the "economics" are less dramatic. After reading an article around the NT2-A, I've been convinced that this was the best solution for my financial state.
Paul White says that this is the mic with less floor noise of the affordable ones, valid for practically anything and, with a very plain response in frequencies. A bright mic that can help to opaque voices, as mine... so... the dices were rolling.
In other side, it's clear that there is a clear difference between a dynamic mic, as the Shure SM57 or SM58, that are designed to catch just the sound very close to its capsule, highly attenuating the sound with the distance and, a large diaphragm mic, as the Rode NT2-A, designed to catch everything around.
Trying to reduce the ambience noises that this mic can catch, I've tried also an absorbent mic screen, concretely, the T.Bone Micscreen.
Follows, my impressions about both things.
Rode NT2-A Studio Solution Set
This nice pack includes the mic NT2-A, the anti-shock mount, an integrated pop filter, an XLR cable, a CD with more marketing than useful info and, a fund for the mic.
First pack that I've received was faulty. The Omni and Eight mode of the mic were working ok related to gain but, they had a lot of floor noise produced by the mic itself, that was far away from that quietness that Paul White was talking about.
The cardiod mode didn't work and the floor noise was really high, with a very weak signal.
So, I did my first tests with the Eight mode and, despite of the floor noise, I was really impressed about how clear and defined were vocals and, best of all, no need to re-EQ the track (except a small dip around 5KHz, to remove some piercing presence).
After a while, I had the replacement pack, with a new mic and... this time was everything working fine!.
Impressive the low level of the floor noise and, impressive the high sensibility of this mic, able to catch the quietest sound happening in the room, even out of the cardioid pattern.
Also, impressive gain level. If my tests with the first faulty mic needed to raise the input gain in the pre-amp, for those tests with the second one, I needed to maintain the gain knob below noon, to avoid the bus to clip.
It's able to suffer high level of sound pressure, accepting 137 dB (normal mode) or, up to 147 dB (with pad set to -10dB). I've started to scream like a monkey asking for an autograph to Cheetah and, the mic get it all, without any distortion. Cool.
Fortunately, the pack include the antishock-mount, since this mic needs to be mount in a ring and, otherwise, I hadn't the opportunity to try it mounted in my mic stand.
The anti-pop filter is just correct and, a bit limited in movements. Possibly, better to buy an anti-pop filter with a flexible arm.
The antishock-mount seems reliable and strong.
The mic weight considerably and has an impressive size. Done to dynamic mics, as the SM57, SM58 and, some Sennheiser, to stand this mic on my hands was a religious experience.
Overall, it looks really professional. It doesn't seems a cheap gear produced at low cost in China and, in fact, Rode' series ending in A are being produced in Australia, where Rode were able to reduce costs by using sophisticated machines that reduced the mounting time and, excess of personal.
The takes I did with this mic (even the faulty one), leaved a track with a nice gain, awesome dynamic range, lot of detail and defined vocals that suited the mix naturally. I had to dip around 5K to remove some excessive bright there but, even without touching the EQ, the result is directly usable.
I am really happy with this mic, indeed.
Due to its high ability to get high preaure level, it's even possible to use it to record guitar or bass amp, and even Kick drums. To do that, we need to switch on the option -10dB pad.
So, a great mic, with an affordable price related to other large diaphragm mics, high quality, impecable finishing, very low floor noise, plain EQ and, the possibilty to use it for any micking task in the Studio.
To whoever that is sick about its SM57 or SM58, I highly recommend him to check this mic!.
T.Bone MicScreen
Well, I've tried to filter part of the ambience sound heard in the back side of the mic (not cardioid zone) and, to avoid room reflexions, by using this mic screen, because it was really affordable and, opinions were very mixed about this kind of solutions.
Comes unmounted but, the task is easy. Everything comes in a box with a very professional look. Has some interesting accessories, as a bar that allows you to mount two mics (for stereo tasks).
The screen as an adjustable wide (but, unstable and not accurate) and, it's easy to couple to any mic stand but, because of its weight, you better choose a stand with a weighty base.
Related to its function... does it really work absorbing the ambience noise?.
Answer is... NOT!.
I've put the stand with the mic facing a books shell (to avoid reflexions) and, with the screen behind, "protecting" the rear side of the mic (not cardioid).
In first place, the NT2-A is cardioid but not hypercardioid.
I've tried to click my fingers in the back zone and, it catches the sound with total clarity, with a volume level slightly lower respect to same sound in front the mic but, high enough.
Then, I've tried the same behind the screen and, the mic was catching the sound without any issues.
Vocals tests were also very clear about all this.
While I was adjusting the gain level, in my PC, far away from the mic and, with headphones on, I was able to hear the sound that I was producing on my desk, while working.
Therefore, in my honest opinion, probably can help to reduce the reflexions that catch the mic on its rear side but, in no way its absorbing the ambiental noise.
In the track, I was able to clearly hear the reflexions in the room (not acoustically treated).
The good thing is that, mounted on the stand and, with that big NT2-A, you seem like an astronaut handling a sophisticated device for spatial exploration and, this can help you to pick up girls. Take a picture of yourself with all this gear and then... forget the screen.
10 April 2013
DIY: Moding the Epiphone Wilshire Ltd - Electronics and pickups
Introduction
Note: this entry was already published in my old Spanish version of this blog, around December/2010. I am just revisiting it here.
This is a good example of mod over an Asian built guitar, to enhance its electronics.
The Epiphone Wilshire Ltd is a guitar with an acceptable wood, good resonance and sustain but, as usually happens to Asian guitars, the weak spot are frets, nut and electronics.
Frets are very tinny so, extreme bendings are being affected and, since their low profile, they will last very few, since you can do very few refreting tasks there.
Electronics components are low quality ones, small pots with metric measurements, etc.
Usually, the soldering job is of bad quality but, not the case in this axe, that had a neat wiring.
In next picture, you can see the removed pickguard. Notice that just the are below pots was shielded but, even that this extends the ground to every electronics component there (except for pickups selector), since the "tap" isn't totally covered, we aren't effectively closing the Faraday's cage created by the electronics cavity. Remember that such a Faraday's cage is there to catch unwanted environmental noises throwing them to ground and avoiding those noises to be caught by electronics components.
Epiphone guitars were mounting bad sounding pickups for a long time but, this guitar mounts some humbuckers designed in USA that, honestly, sound quite nice. Very classic sound, they cut the mix very well, are clear and defined. They have a very low output level, anyway.
When I bought this guitar, I did it thinking on to swap stock pickups later. I was looking for some sound closer to Santana's one for this axe, taking into account that this guitars is some kind of SG but, with more mass so, together with the guitar I've ordered a set of Barekuckle's Abraxas.
Honestly, stock pickups are good enough and there is no need to swap them if you wanted a classic tone.
But I recommend to swap pots with some quality ones (as CTS) and the jack (Swiftcraft).
Rewiring the guitar
First step was to remove strings.
If strings are in good condition and, we have the idea to re-use them, the trick is to leave strings very lose, in a way that we can remove the stop tail piece from their posts, leaving strings and stop tail over the desk, away from our working area.
I we don't want to re-use them and, to mount new brand strings later, the quickest is to cut them.
You should remove stop tail and bridge, because they aren't fixed to the body but, just the strings pressure maintains both on their posts.
We follow removing the pickguard. We should unscrew every screw to remove the pickguard, that will remain linked to the guitar for a few wires (usually, jack hot and ground, pickups wires and bridge ground).
We need then to remove pickups. You should remove pickups' mounting rings. First, totally lose the screws that regulate pickup height, until they fall into the cavity. Later, we can unscrew mounting rings.
We will identify the wires of each pickup and, we will cut the farest extreme (leaving the most of cable length as possible), if we aren't to re-use those electronics components or, you should de-solder those wires if you wanted to re-use those components.
Now that the pickguard is separated from the body, we can remove old pickups and to mount the new ones.
I made a mistake with this guitar. I rely the cavity was already coated with conductive paint (graphite) but, my first try after wiring everything demonstrated the opposite. Be sure to use a multimeter first to determine if there is continuity on the walls. Put your multimeter to read Ohms (a beep is enough) and check if there is continuity between to distant spots of the cavity walls.
If the cavity isn't shielded, is the right moment to spray it with some conductive paint (graphite or copper) or, to recover it with some metallic foils with conductive autoadhesive glue.
In the following picture, we can see the removed pickguard, pickups already removed and, the new Bridge pickup already in place.
To pass pickup conductors across the narrow tunnels of cavity, my trick is to wrap some isolant tape around all wires, shaping the end as a spike, what helps a lot to this task, as you can see in the following picture (both Abraxas already mounted).
As the cavity should be already shielded (not in this case), we need to coat the pickguard with some shielding foils to make this "tap" to effectively close the Faraday`s cage.
In following picture, you can see the back side of the pickguard coated with autoadhesive conductive glue copper foils. Unfortunately, I run out of stock of wide copper foils so, I had to work a lot more with the available narrow foils. With wide foils you will finish the work really fast.
See that the holes for components were covered during the process but, this is not an issue. With a cutter or the head of an screwdriver you can open the big ones. For the smaller (screw holes), it's enough to punch them with a small star-head screwdriver to open the gap.
Next step is to mount components on the pickguard.
Since we are going to change measures from metric to inches, we will need to widen the diameter of pot's holes.
The classy way should be to use a cutting tool able to provide a perfect circle in a quick and fast way (a Dremel, driller or whatever).
The cheesy way is to use some round file to widen that hole.
Since I have not a proper tool, I went the cheesy way.
To ensure the continuity of ground between the cavity walls and the pickguard "tap", you could solder a jumper wire between both. Or you can put some metallic foils inside that overflow a bit on the surface, to be sure that walls and tap are in contact.
In the cavity, over those foils, you can use a screw as the common grounding spot (where to connect bridge ground, shielding ground and any other free ground wire).
Next step is to mount components on the pickguard. After mounting all them, we check the pickguard in place (without screwing it), just to check that they fit in the cavity without issues as they are now.
In this guitar, I've substituted the typical Gibson-like 3-way selector with a 6-way Freeway selector.
If we put that selector in the stock hole, we will see that the pickguard doesn't fits the cavity.
I've had to slightly shift that hole (widening it with a file), until I made the pickguard to fit the cavity.
As potentiometers lay very close, I had to straighten their lugs, to avoid they to enter in contact, because I've mounted each pair (volume and tone) one facing the other, to make the wiring easier.
It's of a great help to have a clear wiring diagram to start our soldering tasks, without going lost.
In this picture, the design I've prepared for this particular axe.
While we wire everything, it's a good idea to highlight with a fluorescent marker, those wires and soldering spots that we have already done.
You should visually check every soldered spot (they should look clean and shinny) and, be sure that no strand is free.
You should also mechanically check the soldered spot. Move the wire circularly and check that it cannot enter in contact with any other lug or component if you bend the wire.
Try to route your wires in a way that they doesn't avoid to easily mount your finished pickguard.
Protect the legs of condensors and any other naked wire (as pickups ground bare wire), with heatshrink tubes or isolant tape, to be sure that they don't accidentally touch an unwanted part of the circuit.
Once we have all components wired on the pickguard, it's time to link the pickguard to the guitar, soldering ickup wires and guitar ground wires. In that way, we could easily work in the pickguard until is strictly necessary to establish the link.
Next picture corresponds to the finished wiring.
Tie wires together, with plastic ties, isolant tape or whatever of use, to avoid they to mess you when mounting the pickguard over the cavity. Note: even that you can cut the length of pickup conductors, I personally prefer to leave them with full length, in prevision that some day can be transferred to a different guitar.
The only pending task is to mount the pickguard, then.
You have now the opportunity to clean your guitar. Clean everything and to nurture that fingerboard if necessary.
We mount the new knobs (inches) that correspond to the new pots (inches). We mount the bridge, stop tail and strings and... time to test the guitar.
Note: this entry was already published in my old Spanish version of this blog, around December/2010. I am just revisiting it here.
This is a good example of mod over an Asian built guitar, to enhance its electronics.
The Epiphone Wilshire Ltd is a guitar with an acceptable wood, good resonance and sustain but, as usually happens to Asian guitars, the weak spot are frets, nut and electronics.
Frets are very tinny so, extreme bendings are being affected and, since their low profile, they will last very few, since you can do very few refreting tasks there.
Electronics components are low quality ones, small pots with metric measurements, etc.
Usually, the soldering job is of bad quality but, not the case in this axe, that had a neat wiring.
In next picture, you can see the removed pickguard. Notice that just the are below pots was shielded but, even that this extends the ground to every electronics component there (except for pickups selector), since the "tap" isn't totally covered, we aren't effectively closing the Faraday's cage created by the electronics cavity. Remember that such a Faraday's cage is there to catch unwanted environmental noises throwing them to ground and avoiding those noises to be caught by electronics components.
Epiphone guitars were mounting bad sounding pickups for a long time but, this guitar mounts some humbuckers designed in USA that, honestly, sound quite nice. Very classic sound, they cut the mix very well, are clear and defined. They have a very low output level, anyway.
When I bought this guitar, I did it thinking on to swap stock pickups later. I was looking for some sound closer to Santana's one for this axe, taking into account that this guitars is some kind of SG but, with more mass so, together with the guitar I've ordered a set of Barekuckle's Abraxas.
Honestly, stock pickups are good enough and there is no need to swap them if you wanted a classic tone.
But I recommend to swap pots with some quality ones (as CTS) and the jack (Swiftcraft).
Rewiring the guitar
First step was to remove strings.
If strings are in good condition and, we have the idea to re-use them, the trick is to leave strings very lose, in a way that we can remove the stop tail piece from their posts, leaving strings and stop tail over the desk, away from our working area.
I we don't want to re-use them and, to mount new brand strings later, the quickest is to cut them.
You should remove stop tail and bridge, because they aren't fixed to the body but, just the strings pressure maintains both on their posts.
We follow removing the pickguard. We should unscrew every screw to remove the pickguard, that will remain linked to the guitar for a few wires (usually, jack hot and ground, pickups wires and bridge ground).
We need then to remove pickups. You should remove pickups' mounting rings. First, totally lose the screws that regulate pickup height, until they fall into the cavity. Later, we can unscrew mounting rings.
We will identify the wires of each pickup and, we will cut the farest extreme (leaving the most of cable length as possible), if we aren't to re-use those electronics components or, you should de-solder those wires if you wanted to re-use those components.
Now that the pickguard is separated from the body, we can remove old pickups and to mount the new ones.
I made a mistake with this guitar. I rely the cavity was already coated with conductive paint (graphite) but, my first try after wiring everything demonstrated the opposite. Be sure to use a multimeter first to determine if there is continuity on the walls. Put your multimeter to read Ohms (a beep is enough) and check if there is continuity between to distant spots of the cavity walls.
If the cavity isn't shielded, is the right moment to spray it with some conductive paint (graphite or copper) or, to recover it with some metallic foils with conductive autoadhesive glue.
In the following picture, we can see the removed pickguard, pickups already removed and, the new Bridge pickup already in place.
To pass pickup conductors across the narrow tunnels of cavity, my trick is to wrap some isolant tape around all wires, shaping the end as a spike, what helps a lot to this task, as you can see in the following picture (both Abraxas already mounted).
As the cavity should be already shielded (not in this case), we need to coat the pickguard with some shielding foils to make this "tap" to effectively close the Faraday`s cage.
In following picture, you can see the back side of the pickguard coated with autoadhesive conductive glue copper foils. Unfortunately, I run out of stock of wide copper foils so, I had to work a lot more with the available narrow foils. With wide foils you will finish the work really fast.
See that the holes for components were covered during the process but, this is not an issue. With a cutter or the head of an screwdriver you can open the big ones. For the smaller (screw holes), it's enough to punch them with a small star-head screwdriver to open the gap.
Next step is to mount components on the pickguard.
Since we are going to change measures from metric to inches, we will need to widen the diameter of pot's holes.
The classy way should be to use a cutting tool able to provide a perfect circle in a quick and fast way (a Dremel, driller or whatever).
The cheesy way is to use some round file to widen that hole.
Since I have not a proper tool, I went the cheesy way.
To ensure the continuity of ground between the cavity walls and the pickguard "tap", you could solder a jumper wire between both. Or you can put some metallic foils inside that overflow a bit on the surface, to be sure that walls and tap are in contact.
In the cavity, over those foils, you can use a screw as the common grounding spot (where to connect bridge ground, shielding ground and any other free ground wire).
Next step is to mount components on the pickguard. After mounting all them, we check the pickguard in place (without screwing it), just to check that they fit in the cavity without issues as they are now.
In this guitar, I've substituted the typical Gibson-like 3-way selector with a 6-way Freeway selector.
If we put that selector in the stock hole, we will see that the pickguard doesn't fits the cavity.
I've had to slightly shift that hole (widening it with a file), until I made the pickguard to fit the cavity.
As potentiometers lay very close, I had to straighten their lugs, to avoid they to enter in contact, because I've mounted each pair (volume and tone) one facing the other, to make the wiring easier.
It's of a great help to have a clear wiring diagram to start our soldering tasks, without going lost.
In this picture, the design I've prepared for this particular axe.
While we wire everything, it's a good idea to highlight with a fluorescent marker, those wires and soldering spots that we have already done.
You should visually check every soldered spot (they should look clean and shinny) and, be sure that no strand is free.
You should also mechanically check the soldered spot. Move the wire circularly and check that it cannot enter in contact with any other lug or component if you bend the wire.
Try to route your wires in a way that they doesn't avoid to easily mount your finished pickguard.
Protect the legs of condensors and any other naked wire (as pickups ground bare wire), with heatshrink tubes or isolant tape, to be sure that they don't accidentally touch an unwanted part of the circuit.
Once we have all components wired on the pickguard, it's time to link the pickguard to the guitar, soldering ickup wires and guitar ground wires. In that way, we could easily work in the pickguard until is strictly necessary to establish the link.
Next picture corresponds to the finished wiring.
Tie wires together, with plastic ties, isolant tape or whatever of use, to avoid they to mess you when mounting the pickguard over the cavity. Note: even that you can cut the length of pickup conductors, I personally prefer to leave them with full length, in prevision that some day can be transferred to a different guitar.
The only pending task is to mount the pickguard, then.
You have now the opportunity to clean your guitar. Clean everything and to nurture that fingerboard if necessary.
We mount the new knobs (inches) that correspond to the new pots (inches). We mount the bridge, stop tail and strings and... time to test the guitar.
09 April 2013
Home Studio: Electric Bass & Guitar direct recording using Pro Tools, Rack 003 and Amplitube
Introduction
Note: this entry was already published in my old Spanish version of this blog around November/2010. I am revisiting it here.
Lucky people will have the opportunity to record their guitars taking the sound of one or more mics directed to their amp's cabs. Less lucky people will have to try to get an acceptable sound by directly recording the guitar, using some kind of amp modeler plugin.
Even if you can mic your amp full kranked, micking technique is complex and, to achieve the best sound is not always so easy. Other techniques, as re-amping, where the clean recorded signal of the guitar is being sent to one or more amps to, finally record back the amplified sound, allows you to work with several mic positions and, several amps and amp' settings until achieve the desired result.
But, this re-amping techniques require good gear (mics, amps, pre-amps) and, you can loose a lot of time moving mics, changing amp settings, etc.
To just record Reference Guitar Tracks, there is nothing as a good amp modeler plugin. You just need to connect your guitar to your audio card, select the right amp in that plugin and take care of the basis sound while recording. If results doesn't convince you, you can later swap the amp model, mics models, speakers models and, whatever other thing that you needed, without having to move yourself out of your chair, until to achieve the wanted sound.
But, sure, amps modelers have their own tricks and limitations.
I'm convinced that Amplitube 3 by IK Multimedia is an excellent simulator, the best in its price range but, I see lot of people that seems to have serious difficulties to get good sound from this plugin. I think you can get decent tone even with Multimedia PC Speakers.
If you are expecting from this plugin to hear the sound of your amp as if you were inside the room where you've put the mics, I think you didn't understand the concept behind Amplitube 3.
Amplitube 3 simulated the recorded sound of an amp that, being catched by a mic, processed by a pre-amp and inserted in a recording track, will be always somewhat different to the "live" sound but, this recording will be very similar to the sound you will hear in any commercial mix.
Evidently, the near field monitors of your home studio aren't able to move the amount of air that an amp cab can move and, therefore, the sensation produced by being in front of your amp disappears and, the sound becomes a bit more compressed, dry and colored by the mic (which position and direction highly affects the recorded sound) and pre-amp.
Studio, Line, Instrument, Mic levels and other stuff
While testing a DI box (see previous blog) I've realized that the input signal was highly superior to the input level I was achieving through other paths, what made me to review the links between my gear, that was previously well configured but, that I've messed up when changing to a new home.
A perfect coupling between the outputs of a device and the inputs of the following one in the chain is of high importance to achieve a clean, strong and free of distortion signal. But.... there are so many types of signal levels!.
Studio level
Gear that processes audio signals (pre-amps, compressors, equalizers, etc) can have inputs or outputs ready for Studio Level signals. The Studio level is the highest signal level of all (+4 dBu).
If we plug a Studio Level output in a input with an inferior level, that high signal will overflow the input, creating a high distortion level and forcing the input to clip the sound.
Usually, Studio Level signals travel thru a cable with connectors type XLR or TRS and, usually they have balanced signals. Impedance is usually around 600 Ohm.
A balanced signal consist into send twice the signal but, with opposite phase to cancel noise.
If your equipment is stereo and has two balanced XLR or TRS outputs (one by channel), they will possibly be a couple of Studio Level outputs.
Line Level
The Line Level is clearly inferior to Studio Level (-10 dB) and, this is the level where the consumer electronics devices usually work (Hi Fi chains, etc.). Apart of such a level, the needs respect of impedances are totally different from those that are required for a mic or an electric instrument (guitar, bass).
Usually, this signal Level travels thru cables with TS (Jack mono 1/4") or TRS (stereo, if it comes from an stereo device) and, they are unbalanced (left channel, right channel and ground).
The impedance level is around 10 KOhm.
Instrument Level
Electric instruments, as the guitar or bass, generate a very low signal but, stronger than a mic. For sure, that signal should be amplified (that's why we use guitar amps!) and, its level is clearly under the Line Level.
Additionally, instruments need of inputs with high impedance levels and, outputs with very low impedance level to achieve their best dynamic range.
The instrument signal usually travel in the same way as the Line signals, trough a cable with Jacks TS (usually) or TRS (if an stereo instrument) and, they are unbalanced.
The input impedance for an instrument should be over the 20 KOhm and, usually around 1 MOhm.
Mic Level
Mics generate very weak signal levels, the weakest of all. The mic needs of an input impedance level lower than the Line or Instrument levels and, needs to be HIGHLY amplified.
Mic signals usually travel through cables with XLR connectors and, they are balanced.
Additionally, some mic types requiere Phantom Power so, they would need an input that can provide those 48V that are a must for the mic to produce some signal (by example, condensor mics).
Impedance level is around 600 Ohm.
This kind of signal is one of the most used in Studio. The reason is that their technical characteristics allow to send signals in longer distances with less lost of quality of such a signal.
In instrument cables, you can notice some degradation around 10 meters (depending on the quality of cable and connectors).
Sample of Signal Level pairing using the Rack 003
Rack 003 inputs from 1 to 4 are inputs with pre-amp. That means that they will accept weak signals that should be lately amplified.
Each of these inputs has to connectors, one XLR connector and one TSR, both balanced (the device will unbalanced them, depending on what are you pluggin in there).
The XLR input is waiting for a Mic Level signal, with the right level and impedance used for such a signal. To connect there any other kind of signal will produce a highly distorted signal, due to the fact that the output gain in this input is the higher one.
In Rack 003, each couple of inputs (1 & 2 and, 3 & 4) have some switch to activate the phantom power. The drawback is that once you switch it on, both inputs have the phantom power active and, this will mean that you cannot link there any equipment that doesn't requires Phantom Power. Be careful, then.
So... what do I have linked to the pair of XLR inputs 1 and 2?.
You can not simultaneously use the XLR and DI inputs, even that you can leave the equipment linked there, just one of the two input paths will be active at once.
We will select the input type with the button "Mic/DI" that you will find in the front panel of your Rack (mic = off, DI = light). If the led of that button blinks, this means that this input is clipping and therefore, the level of the signal that you are providing there is higher than foreseen.
What do I have connected to inputs 3 and 4?
A button in the back panel close to each entry allows you to select the Line level (-10dB, button pushed) or Studio Level (+4dB, button pulled).
What do I have in inputs 5 to 8?
Carefully check technical specifications related to inputs and outputs of all your gear to be sure that your are properly matching each output to the right input and viceversa.
This can sound obvious but, this creates me a lot of headaches and disappointing results in the past.
What's a good input level?
Mi own experience with Pro Tools and Amplitube 3 say that it's better to have to highest signal level in your input track and, then to regulate the output level with the Master knob of Amplitube 3.
The higher your input signal level, the higher the difference between signal and floor noise and, the best the different amp models and stomp boxes will react in Amplitube 3. A very weak signal makes this plugin to sound really awful.
The average sound (RMS) should be between the yellow zone of the input meter and, peaks should be under the red area but, close to its border. Strum hard chords to control peaks level.
Even that Amplitube 3 has a gain knob (input) and, that knob allows to you to raise the input level inside the amp simulator, results are way worst than leaving the knob in its default position and, providing a higher input level.
Reason is that a weak signal has a very low dynamic range and, signal and floor noise are very close in loudness so, increasing the gain to this kind of signal is also increasing the gain of the floor noise, what results in a noisy sound, noise that is being increased to higher levels by the several amp models and gain stomp boxes.
This is true for every input signal in Pro Tools. It's better to send the input track to an auxiliary track and, to lower the output volume in such a track (after being processed by other plugins or outboard gear).
Searching for the right amp' sound
First step will be to select one of the available amps, depending on your target sound and, tweak the tone stack, gain and volume to taste.
Since the input volume will be high, the amplified volume will be really higher and, therefore, your auxiliary track will start clipping. It's better to lower the output volume of this plugin by using the Master Knob, without reduce the rest of controls of your amp model. Since we are in Digital Audio world, your peaks should be maintained below -3dB, to avoid clipping the input of the next device/plugin/track in the chain.
Hum and rest of candies
Amplitube 3 are modeling really well the originals, so good that they even model the natural noise of the modeled gear. Therefore, if the original is noisy, your model will be also.
Guitars, and very specially those loaded with single coils, are introducing their own noises and hum. The noise generated by the own amp model couldn't be avoided but, you can get rid of your guitar noises in several ways.
You can use some Noise Gate or Noise Reduction pedal between your guitar and the Rack 003 (as the ISP Decimator G-String) or, you can rely in the Noise Gate that provides the own Amplitube 3 plugin or, you can insert a Noise Gate plugin in your auxiliary track, just before the Amplitube 3 plugin.
To check that you are effectively reducing the guitar noise, put Amplitube in Bypass mode, to be sure that you aren't mixing amp and guitar noises.
Any kind of Gate you can use there, can have its own drawbacks and, you can even ruin your guitar sound if your weaker sounds are very close the the floor noise. To accurately adjust a Gate can be a real headache.
Sometimes, it's better to reduce the noise at the beginning and end of the track and, during the "void" parts or, better, deleting "no info" parts or, doing an automate fade to zero in such a sections (where there is only noise).
This is a concept that many people doesn't realizes about, tube amplifies have their own floor noise. Being Amplitube 3 a great emulator, it will reproduce also such a characteristic noise.
Adjusting the Gain Level
Some of amp models are vintage amps. Vintage amps were more oriented to achieve a clean sound than other thing. By example, the power of an amp meant the power that would provide while standing clean (so, 50W meant, 50W clean but, some more watts after beginning to distort!).
Since the goal was to keep the amp clean at high power levels, you will need to help such a kind of amps to break up with the help of some stomp boxes.
Typically, you used some overdrive, fuzz or distortion pedals to force those tubes to start to distort.
You will need to do the same in Amplitube 3. Some vintage models can sound really cold until you insert some gain pedal before (overdrive, fuzz. booster...).
So, if you feel that your amp lacks some guts, try first some typical overdrive (Tubescreamer) or, some typical distortion (RAT), to get the sweet spot of those "virtual tubes". It works really well.
Once, you've achieved a nice tone, lower the output signal of this plugin with the help of the Master Knob (don't modify your volume or gain). This will allow to further plugins to have some headroom to process the dynamics of this output signal. My recommendation is that the output RMS will be around -20 dB. You will push the complete mix later, with the help of some compressor in the mix buss.
Fine tuning the sound: cabs and mics emulation
This is probably the modules of Amplitube that more evoluted respect of previous versions of this plugin.
The possibility to change your cab, mics and spatial position of mics, brings to you as many possibilities to experiment with the sound as you would have in a real Studio but, without having to leave your chair!.
You can try swapping the cab model. Maybe, you will like more the sound of such an amp paired with different speakers. It always depends on what are you after.
The mic type and, its position, together with the distances to speakers and to other mics have a lot of impact in the sound. Just check different mics for each position, one at once and, play with the respective position of each mic, separately and together.
The rule of 3 applies as in the real world. To avoid phase issues, one mic should be place 3 times to the distance of the the other, respect of the source of the sound. But, for sure, you can play with any position and, maybe some cancellation of phase is just what you were looking for.
Once you have positioned the main mics, you can concentrate in the ambience mics. You can only control here the gap between both mics and, the volume with which this mics will be blended with the main mics.
Don't be lazy trying several possibilities. You have no physical effort, tests are easy and fast, with immediate results. And, the best of all, once you keep your guitar clean take, you can change everything at any time achieving totally different results.
Adding Effects
Not a good idea to start checking stompboxes (other than those that just put the tubes in their sweet spot) before having achieved your foundational amp tone.
Then you can check the stomp boxes and, try as much as you liked trying to get the exact sound you are after. Those stomp boxes are really well modeled and behave really close to the real thing.
Amplitube 3 gives you the opportunity to swap the order in your chain of pedals so, you can check the effect of each pedal before or after others and, choose the combination that better works for you. Everything, without changing a cable or moving a pedal physically so... play with that!. And, remember that if you don't like the results at the end, you will be able to change everything next day.
Note: guitar amplifiers tend to be recorded WITHOUT amp's reverberation. The reverb effect (as well as the delay effect) is usually added in the mix, by using some quality plugin to give the right deep to your mix.
The good thing of Amplitube3 is that with the same recorded clean track, you can try for ever up to get what you wanted. You just need a single good performance over which work later.
Cutting the Mix
Once you fine tuned the sound of your guitar/amp/effects and achieved a nice sounding guitar track alone, it's possible that your track cannot cut the mix with authority.
In previous blog entries, I've discussed about the 3 dimensions of the Mixing process. Please read those entries to understand how to equalize the guitar and rest of instruments, in a way that every instrument is well represented in the stereo image.
EQ usually goes after Amplitube, to enhance or dismiss wanted frequencies, once your foundational tone was achieved.
Giving strength and consistence to guitar' sound
Even not being always strictly necessary, a light processing of the guitar sound, through a compressor can work in several ways: leveling peaks and valleys, providing a more homogeneous and louder volume, adding some touch of color, modify dynamics, adding or resting punch, increasing sustain, etc. We discussed about it in previous blog entries also.
My personal preference for guitars is a light touch of the plugin Fairchild 670 by IK Multimedia. I am more interested on the coloration that gives that plugin to the sound than on a pure compression.
For basses, it's very usual to combine the LA-2A and 1176 plugins but, I also like the results of the Dynamics plugin by Sonnox.
You can increase the "meat" of the instrument' sound, adding one more plugin that can enhance the harmonical content, like the BBE or, Inflator by Sonnox, among other harmonics exciters.
You could use a colorant equalizer also, as the Pulteq EQP-1A, at the end, to enhance bass frequencies to the bass guitar and give more punch.
At the end, the chain of sound processors that you will choose for your particular project can be so variate as your imagination.
Note: this entry was already published in my old Spanish version of this blog around November/2010. I am revisiting it here.
Lucky people will have the opportunity to record their guitars taking the sound of one or more mics directed to their amp's cabs. Less lucky people will have to try to get an acceptable sound by directly recording the guitar, using some kind of amp modeler plugin.
Even if you can mic your amp full kranked, micking technique is complex and, to achieve the best sound is not always so easy. Other techniques, as re-amping, where the clean recorded signal of the guitar is being sent to one or more amps to, finally record back the amplified sound, allows you to work with several mic positions and, several amps and amp' settings until achieve the desired result.
But, this re-amping techniques require good gear (mics, amps, pre-amps) and, you can loose a lot of time moving mics, changing amp settings, etc.
To just record Reference Guitar Tracks, there is nothing as a good amp modeler plugin. You just need to connect your guitar to your audio card, select the right amp in that plugin and take care of the basis sound while recording. If results doesn't convince you, you can later swap the amp model, mics models, speakers models and, whatever other thing that you needed, without having to move yourself out of your chair, until to achieve the wanted sound.
But, sure, amps modelers have their own tricks and limitations.
I'm convinced that Amplitube 3 by IK Multimedia is an excellent simulator, the best in its price range but, I see lot of people that seems to have serious difficulties to get good sound from this plugin. I think you can get decent tone even with Multimedia PC Speakers.
If you are expecting from this plugin to hear the sound of your amp as if you were inside the room where you've put the mics, I think you didn't understand the concept behind Amplitube 3.
Amplitube 3 simulated the recorded sound of an amp that, being catched by a mic, processed by a pre-amp and inserted in a recording track, will be always somewhat different to the "live" sound but, this recording will be very similar to the sound you will hear in any commercial mix.
Evidently, the near field monitors of your home studio aren't able to move the amount of air that an amp cab can move and, therefore, the sensation produced by being in front of your amp disappears and, the sound becomes a bit more compressed, dry and colored by the mic (which position and direction highly affects the recorded sound) and pre-amp.
Studio, Line, Instrument, Mic levels and other stuff
While testing a DI box (see previous blog) I've realized that the input signal was highly superior to the input level I was achieving through other paths, what made me to review the links between my gear, that was previously well configured but, that I've messed up when changing to a new home.
A perfect coupling between the outputs of a device and the inputs of the following one in the chain is of high importance to achieve a clean, strong and free of distortion signal. But.... there are so many types of signal levels!.
Studio level
Gear that processes audio signals (pre-amps, compressors, equalizers, etc) can have inputs or outputs ready for Studio Level signals. The Studio level is the highest signal level of all (+4 dBu).
If we plug a Studio Level output in a input with an inferior level, that high signal will overflow the input, creating a high distortion level and forcing the input to clip the sound.
Usually, Studio Level signals travel thru a cable with connectors type XLR or TRS and, usually they have balanced signals. Impedance is usually around 600 Ohm.
A balanced signal consist into send twice the signal but, with opposite phase to cancel noise.
If your equipment is stereo and has two balanced XLR or TRS outputs (one by channel), they will possibly be a couple of Studio Level outputs.
Line Level
The Line Level is clearly inferior to Studio Level (-10 dB) and, this is the level where the consumer electronics devices usually work (Hi Fi chains, etc.). Apart of such a level, the needs respect of impedances are totally different from those that are required for a mic or an electric instrument (guitar, bass).
Usually, this signal Level travels thru cables with TS (Jack mono 1/4") or TRS (stereo, if it comes from an stereo device) and, they are unbalanced (left channel, right channel and ground).
The impedance level is around 10 KOhm.
Instrument Level
Electric instruments, as the guitar or bass, generate a very low signal but, stronger than a mic. For sure, that signal should be amplified (that's why we use guitar amps!) and, its level is clearly under the Line Level.
Additionally, instruments need of inputs with high impedance levels and, outputs with very low impedance level to achieve their best dynamic range.
The instrument signal usually travel in the same way as the Line signals, trough a cable with Jacks TS (usually) or TRS (if an stereo instrument) and, they are unbalanced.
The input impedance for an instrument should be over the 20 KOhm and, usually around 1 MOhm.
Mic Level
Mics generate very weak signal levels, the weakest of all. The mic needs of an input impedance level lower than the Line or Instrument levels and, needs to be HIGHLY amplified.
Mic signals usually travel through cables with XLR connectors and, they are balanced.
Additionally, some mic types requiere Phantom Power so, they would need an input that can provide those 48V that are a must for the mic to produce some signal (by example, condensor mics).
Impedance level is around 600 Ohm.
This kind of signal is one of the most used in Studio. The reason is that their technical characteristics allow to send signals in longer distances with less lost of quality of such a signal.
In instrument cables, you can notice some degradation around 10 meters (depending on the quality of cable and connectors).
Sample of Signal Level pairing using the Rack 003
Rack 003 inputs from 1 to 4 are inputs with pre-amp. That means that they will accept weak signals that should be lately amplified.
Each of these inputs has to connectors, one XLR connector and one TSR, both balanced (the device will unbalanced them, depending on what are you pluggin in there).
The XLR input is waiting for a Mic Level signal, with the right level and impedance used for such a signal. To connect there any other kind of signal will produce a highly distorted signal, due to the fact that the output gain in this input is the higher one.
In Rack 003, each couple of inputs (1 & 2 and, 3 & 4) have some switch to activate the phantom power. The drawback is that once you switch it on, both inputs have the phantom power active and, this will mean that you cannot link there any equipment that doesn't requires Phantom Power. Be careful, then.
So... what do I have linked to the pair of XLR inputs 1 and 2?.
- Phantom Power is active
- In input 1 I'm directly pluging mics that require phantom power (condenser mics)
- In input 2 I'm connecting the output of the Radial J48 Active DI, that converts the Instrument Level signal of its input into a Balanced Mic output that's being sent thru its output XLR connector.
You can not simultaneously use the XLR and DI inputs, even that you can leave the equipment linked there, just one of the two input paths will be active at once.
We will select the input type with the button "Mic/DI" that you will find in the front panel of your Rack (mic = off, DI = light). If the led of that button blinks, this means that this input is clipping and therefore, the level of the signal that you are providing there is higher than foreseen.
What do I have connected to inputs 3 and 4?
- I am directly plugin any kind of instrument to input 3 DI, manipulating the gain level labeled as "Input 3" in the front end of the rack.
- To input 4, I've plugged the output FANTA of the TAD Silencer. This output with speaker emulation of this attenuator/load box has a very low level, as a mic, is balanced and works with an XLR connector. Since it doesn't needs Phantom Power, I am connecting it to input 4, instead of 1 or 2 (with phantom power) to avoid damaging the unit. The gain level of this input can be regulated with the gain knob labeled "Input 4" in the front panel of the rack.
A button in the back panel close to each entry allows you to select the Line level (-10dB, button pushed) or Studio Level (+4dB, button pulled).
What do I have in inputs 5 to 8?
- To input 5 I've connected the LINE output of the Silencer. It has a Line Level (-10dB) and, the gain for this signal is being controlled in the Silencer itself, with the volume control of the line output.
- To input 6, I've connected the balanced output of the SPL Track One (pre-amp). It provides an Studio Level output (+4 dBu), with an XLR connector, converted to TSR in the input side.
Carefully check technical specifications related to inputs and outputs of all your gear to be sure that your are properly matching each output to the right input and viceversa.
This can sound obvious but, this creates me a lot of headaches and disappointing results in the past.
What's a good input level?
Mi own experience with Pro Tools and Amplitube 3 say that it's better to have to highest signal level in your input track and, then to regulate the output level with the Master knob of Amplitube 3.
The higher your input signal level, the higher the difference between signal and floor noise and, the best the different amp models and stomp boxes will react in Amplitube 3. A very weak signal makes this plugin to sound really awful.
The average sound (RMS) should be between the yellow zone of the input meter and, peaks should be under the red area but, close to its border. Strum hard chords to control peaks level.
Even that Amplitube 3 has a gain knob (input) and, that knob allows to you to raise the input level inside the amp simulator, results are way worst than leaving the knob in its default position and, providing a higher input level.
Reason is that a weak signal has a very low dynamic range and, signal and floor noise are very close in loudness so, increasing the gain to this kind of signal is also increasing the gain of the floor noise, what results in a noisy sound, noise that is being increased to higher levels by the several amp models and gain stomp boxes.
This is true for every input signal in Pro Tools. It's better to send the input track to an auxiliary track and, to lower the output volume in such a track (after being processed by other plugins or outboard gear).
Searching for the right amp' sound
Once we've achieved a high and clean input signal, this is time to choose the amp model and to tweak controls until getting the wanted tone.
I'm always recording a clean track of the guitar sound, in a mono track that I'm sending to an auxiliary stereo track. This is where I insert that plugins that directly affect to the guitar' sound and, therefore, where Amplitube 3 is being inserted (usually as the first plugin).
First step will be to select one of the available amps, depending on your target sound and, tweak the tone stack, gain and volume to taste.
Since the input volume will be high, the amplified volume will be really higher and, therefore, your auxiliary track will start clipping. It's better to lower the output volume of this plugin by using the Master Knob, without reduce the rest of controls of your amp model. Since we are in Digital Audio world, your peaks should be maintained below -3dB, to avoid clipping the input of the next device/plugin/track in the chain.
Hum and rest of candies
Amplitube 3 are modeling really well the originals, so good that they even model the natural noise of the modeled gear. Therefore, if the original is noisy, your model will be also.
Guitars, and very specially those loaded with single coils, are introducing their own noises and hum. The noise generated by the own amp model couldn't be avoided but, you can get rid of your guitar noises in several ways.
You can use some Noise Gate or Noise Reduction pedal between your guitar and the Rack 003 (as the ISP Decimator G-String) or, you can rely in the Noise Gate that provides the own Amplitube 3 plugin or, you can insert a Noise Gate plugin in your auxiliary track, just before the Amplitube 3 plugin.
To check that you are effectively reducing the guitar noise, put Amplitube in Bypass mode, to be sure that you aren't mixing amp and guitar noises.
Any kind of Gate you can use there, can have its own drawbacks and, you can even ruin your guitar sound if your weaker sounds are very close the the floor noise. To accurately adjust a Gate can be a real headache.
Sometimes, it's better to reduce the noise at the beginning and end of the track and, during the "void" parts or, better, deleting "no info" parts or, doing an automate fade to zero in such a sections (where there is only noise).
This is a concept that many people doesn't realizes about, tube amplifies have their own floor noise. Being Amplitube 3 a great emulator, it will reproduce also such a characteristic noise.
Adjusting the Gain Level
Some of amp models are vintage amps. Vintage amps were more oriented to achieve a clean sound than other thing. By example, the power of an amp meant the power that would provide while standing clean (so, 50W meant, 50W clean but, some more watts after beginning to distort!).
Since the goal was to keep the amp clean at high power levels, you will need to help such a kind of amps to break up with the help of some stomp boxes.
Typically, you used some overdrive, fuzz or distortion pedals to force those tubes to start to distort.
You will need to do the same in Amplitube 3. Some vintage models can sound really cold until you insert some gain pedal before (overdrive, fuzz. booster...).
So, if you feel that your amp lacks some guts, try first some typical overdrive (Tubescreamer) or, some typical distortion (RAT), to get the sweet spot of those "virtual tubes". It works really well.
Once, you've achieved a nice tone, lower the output signal of this plugin with the help of the Master Knob (don't modify your volume or gain). This will allow to further plugins to have some headroom to process the dynamics of this output signal. My recommendation is that the output RMS will be around -20 dB. You will push the complete mix later, with the help of some compressor in the mix buss.
Fine tuning the sound: cabs and mics emulation
This is probably the modules of Amplitube that more evoluted respect of previous versions of this plugin.
The possibility to change your cab, mics and spatial position of mics, brings to you as many possibilities to experiment with the sound as you would have in a real Studio but, without having to leave your chair!.
You can try swapping the cab model. Maybe, you will like more the sound of such an amp paired with different speakers. It always depends on what are you after.
The mic type and, its position, together with the distances to speakers and to other mics have a lot of impact in the sound. Just check different mics for each position, one at once and, play with the respective position of each mic, separately and together.
The rule of 3 applies as in the real world. To avoid phase issues, one mic should be place 3 times to the distance of the the other, respect of the source of the sound. But, for sure, you can play with any position and, maybe some cancellation of phase is just what you were looking for.
Once you have positioned the main mics, you can concentrate in the ambience mics. You can only control here the gap between both mics and, the volume with which this mics will be blended with the main mics.
Don't be lazy trying several possibilities. You have no physical effort, tests are easy and fast, with immediate results. And, the best of all, once you keep your guitar clean take, you can change everything at any time achieving totally different results.
Adding Effects
Not a good idea to start checking stompboxes (other than those that just put the tubes in their sweet spot) before having achieved your foundational amp tone.
Then you can check the stomp boxes and, try as much as you liked trying to get the exact sound you are after. Those stomp boxes are really well modeled and behave really close to the real thing.
Amplitube 3 gives you the opportunity to swap the order in your chain of pedals so, you can check the effect of each pedal before or after others and, choose the combination that better works for you. Everything, without changing a cable or moving a pedal physically so... play with that!. And, remember that if you don't like the results at the end, you will be able to change everything next day.
Note: guitar amplifiers tend to be recorded WITHOUT amp's reverberation. The reverb effect (as well as the delay effect) is usually added in the mix, by using some quality plugin to give the right deep to your mix.
The good thing of Amplitube3 is that with the same recorded clean track, you can try for ever up to get what you wanted. You just need a single good performance over which work later.
Cutting the Mix
Once you fine tuned the sound of your guitar/amp/effects and achieved a nice sounding guitar track alone, it's possible that your track cannot cut the mix with authority.
In previous blog entries, I've discussed about the 3 dimensions of the Mixing process. Please read those entries to understand how to equalize the guitar and rest of instruments, in a way that every instrument is well represented in the stereo image.
EQ usually goes after Amplitube, to enhance or dismiss wanted frequencies, once your foundational tone was achieved.
Giving strength and consistence to guitar' sound
Even not being always strictly necessary, a light processing of the guitar sound, through a compressor can work in several ways: leveling peaks and valleys, providing a more homogeneous and louder volume, adding some touch of color, modify dynamics, adding or resting punch, increasing sustain, etc. We discussed about it in previous blog entries also.
My personal preference for guitars is a light touch of the plugin Fairchild 670 by IK Multimedia. I am more interested on the coloration that gives that plugin to the sound than on a pure compression.
For basses, it's very usual to combine the LA-2A and 1176 plugins but, I also like the results of the Dynamics plugin by Sonnox.
You can increase the "meat" of the instrument' sound, adding one more plugin that can enhance the harmonical content, like the BBE or, Inflator by Sonnox, among other harmonics exciters.
You could use a colorant equalizer also, as the Pulteq EQP-1A, at the end, to enhance bass frequencies to the bass guitar and give more punch.
At the end, the chain of sound processors that you will choose for your particular project can be so variate as your imagination.
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