Review: Marshall JCM 2000 DSL

Hello and welcome to this week's article!

Today we're going to review one of the most classic Marshall Heads: the JCM 2000 DSL.

This version came out in 1997 and it was meant to be as a dual-channel "expanded version" of a Jcm 800 and 900, but it turned out to be a completely different beast.

DSL means Dual Super Lead and indicates the fact that the signal passes through two gain stages, so the added gain is more than a single stage (there are heads with up to six stages, like the Peavey 5150, though), and there is also a version called TSL, Triple Super lead, which is supposed to provide even more gain (and a third channel).

This head has been used, through the years from some of the biggest bands around, ranging from rock (for example Muse), to thrash metal (Trivium).

The head has a nice sound, with the mids typical of a Marshall, but yet the Tone Shift and Deep control lets you have enough versatility to obtain enough different tones.

Around the web there is a rumor that for some reason a Jcm 800 has a thicker, more powerful sound than a Jcm 2000 Dsl, and even more a Tsl (which has been wildly criticized for this reason), and this could be true, but I actually find the Jcm 2000 to be a nice halfway between the classic Marshall tone, the versatility required by today's musicians, and a tone that is more manageable at lower volumes (while, notoriously, a Jcm 800 needs to be cranked to the max in order to unleash its full potential).

Today Marshall is selling an updated version of the Jcm 2000 called only Dsl 100, which is identical to the original version except for some interface improvement: 

- A Resonance control instead of the Deep Switch

- A versatile Digital Reverb instead of the old spring one

- A pentode/triode switch that lets you choose between 100 and 50w.

Specs taken from the website:

- Two channels - Classic Gain and Ultra Gain

- Lead 1 to Lead 2 switching on the Ultra Gain channel

- Tone Shift switch to reconfigure the way the tone section (particularly the Middle control) works

- Deep Switch for increased bottom end

- Effects Loop with level selection switch

- Output (RMS): 100 Watts

- Number of Channels: 2

- Modes Per Channel: 2

- Spring Reverb

- Preamp Tubes: 4 x ECC83

- Power Amp Tubes: 4 x EL34

- Dimensions (approximate): 29.5 x 11.7 x 8.43" (748 x 297 x 214mm)

- Weight (Kg): 49.5 lbs (22.5 kg)

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Acoustic Guitar Vst Simulators (free Vst Plugins)

Hello everyone and welcome to this week's article!
Today we're going to talk about acoustic guitar simulators, which are those plugins that takes the sound of our electric guitar recorded straight into the audio interface, and try to turn it into the sound of an acoustic guitar.

This article implies obviously that we don't have an actual acoustic guitar, nor ways to record it with microphones, which actually would be the best way to track it, so let's just focus on how to turn the sound of our electric guitar into the one that resembles an acoustic one!
Someone can also use this method to use his electric guitar to play some part which would be too unconfortable to be played with an acoustic one, and inserting them into an acoustic sounding context.

Usually plugins that tries to replicate the sound of an acoustic guitar works by modifying the eq, the transient, the resonance and other parameters of the original sound, or by applying an IR impulse, to recreate the resonance of a real acoustic guitar (Click Here for a vide made by Chandler Guitar, which also contains an acoustic guitar IR impulse to download).
Once we have turned the sound of our guitar into a sound that resembles an acoustic one, we can mix it as we would treat the sound of a real acoustic one, and the result sometimes can be strikingly resemblant.


Here are some interesting plugins, some free and some paid, to simulate an acoustic guitar sound:


Voxengo Boogex: a classic free IR impulse loader, that if we use with specific acoustic guitar impulses can really turn our electric guitar sound into an acoustic one.

Nusoft Deepboard: an interesting, small and free plugin that turns the sound of our electric guitar into an acoustic one

BodiLizer: a paid (30$) acoustic simulator which uses an internal IR impulse.

Positivegrid Jamup: a great paid plugin, probably the best in the market today, which has also a good acoustic simulation

Waves Maserati Acg: another paid plugin, designed by the producer Tony Maserati, which has a set of tools for shape the acoustic guitar sound



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Review: Seymour Duncan Pegasus and Sentient Set (with sample)

Hello everyone and welcome to this week's article!
This time we will talk about two recent pickups produced by Seymour Duncan, originally designed only for seven strings guitars, but that today are offered in the 8-strings and 6-strings variants too.

This is a set of a Bridge and a Neck humbucking pickups, with the neck one (the Sentient) being described from the manufacturer as a half-way among a '59 and a Jazz model, so a pickup with a not so high output, but with a nice, round tone, very good with overdrives, and with great dynamics.
An ideal pickup for cleans, and moderately distorted solos.
The bridge pickup instead is a higher output humbucker, called Pegasus, with an Alnico V magnet (like the Sentient), but that distinguish itself from the other 7 strings metal pickup produced by Seymour Duncan, the Nazgul.

While the Nazgul is a very high output pickup, Invader style, with a lot of bass and a chunky tone which resembles the active ones, ideal for extreme metal, the Pegasus has a totally different flavour:
it has a mid-focused tone, due to the Alnico V magnet, and a bit lower output (and therefore bass frequences, which are the ones who becomes more prominent with the increase of the output), and it is created for progressive rock, progressive metal and djent guitarists.

Why did Seymour Duncan feel the need to produce a different pickup suited for the needs of these guitar players? Because Alnico magnets (those mounted usually on Gibson pickups too) have a less violent and more musical (and a little more vintage) feel to the tone than the ceramic ones (used on the Nazgul), and those pickups are created yes to manage heavily distorted sounds, but also to mantain clarity when playing distorted arpeggios or single strings;
clarity and tightness are basilar when playing complex, distorted riffs, so the output and the frequency range have been adjusted (lowering them) to make everything more intelligible.

The Sentient and Pegasus pickups comes in 3 versions: the passive mount with uncovered coils, the passive mount with covered coils and the active mount, which is meant to replace active pickups with a shape similar to theirs.

A sample of Sentient (on the clean) and Pegasus (on the distorted part) can be heard on our Emissary Vst page.



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HOW TO MIX A LIVE BAND (a guide for dummies) 4/4

Hello and welcome to this last part of our super basic tutorial about how to mix a live band in a small gig!

In the latest part we have seen basically every aspect of a mixer, every aspect but the Aux Sends one, which we are going to analyze today.
The Auxiliary Send (Aux Send) is an additional output of the mixer that lets us do another mix, different by the one that goes into the PA, and that we can use for other purposes, for example to send it to the stage monitors.

The band in the stage plan example on the first part of our article has 3 monitors: one for the drummers and 2 for the other band members, covering the left and the right side of the stage.
in the picture above we have 2 aux sends, but usually 12 channels mixers have 3 of them.
So how do we do our relative mix? If we have the aux pots on each channel at noon, the guys on stage will hear by the monitors basically the same mix that can be heard outside, but usually they will want a different mix, because there are things that they need to hear to play better, and others which are less useful and can be eliminated, plus keep in mind that usually with cheap mixers and monitors it's better to send a maximum of 3 or 4 channels per monitor, to make sure the monitor will reproduce them loud and clear without distorting.


An example of monitor mix, always referring to the band in the first part of this tutorial, could be the following:

Aux 1: Left monitor (as seen from the point of view of the drummer): Vocals, keyboards, snare, kick.

Aux 2: Drummer monitor: Vocals, guitar, keyboards, bass

Aux 3: Right monitor: Vocals, the other guitar (or the keyboard), snare, kick

in order to obtain this we will close all aux knobs on the mixer, then open just the Aux 1 pot on Vocal, Keyboard, Snare and Kick, choosing the level according to the volume they need,
then repeat the same procedure with the Aux 2 Pot and the Aux 3, sending only the requested channels to each monitor.
Usually during the soundcheck the sound engineer asks the band to play a part of a song and adjusts the aux sends accordingly, until everyone can hear everything decently.
Once the stage monitors are set in a stable way, the sound engineer can make the last touches to balance everything, keeping in mind that additional modifications will need to be made when the room will be full of people, because the venue responds differently according to the people inside.

Some last suggestion about how to mix a live band:

- If present, the High Pass Filter button is a good way to clean up the low end from unwanted resonating frequences. It can be used basically on each channel, with the exception of Bass and Kick Drum.

- Try to push each sound with the eq slighly towards its place in the mix, for example you can lower a bit the lows on a guitar track and add some grit on the highs, or carve some mids out of toms, remove some highs on a bass track or boost some mids on vocals, but don't exaggerate, and remember that a sligh cut sounds usually better than a sligh boost.



Additional ways to use an Aux Send:

- It can be used to go to external processors, such as a Compressor, or an Fx unit, and then the signal can be routed to the Aux in (sometimes called Aux Return) to be blended with the original one.

- It can be used also to go to a recording machine, like a computer, if there aren't other dedicated outs in the mixer.



Well, the tutorial is over, I hope it was helpful, leave comments below if you have some other tip to share!


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HOW TO MIX A LIVE BAND (a guide for dummies) 3/4

Hello and welcome to the third part of this week's article!

Last week we have seen how to make the gain staging for microphone inputs, today we will talk about the last 2 channels of  our mixer, that we have reserved for "keyboards - d.i.".

If we have a keyboard, it's much likely that it will have a left and right output, so we must connect them to a d.i. box to balance the signal, and from there go into the channel box, this way we will be able to control both left and right channels on the mixer, but if the keyboard player has more outputs, he will need an additional mixer to connect all of them, make a "keyboard submix", and from the stereo output go to the d.i. box, and from them to the channel box and to our mixer.

Since we have only 2 channels for quite a wide array of things left, we need to be creative: if the band has some pre-recorded base we should treat it the same way we could treat keyboards, or if they use both things, we will have to dedicate a single mono channel to keyboards and the other one to the base. The gain staging procedure it's the same explained in the part 2 of this guide, with the addition that usually d.i. boxes have a db attenuation switch, if the signal it's too loud, and a ground/lift switch, to move if there is some hum.

Notice that often the last channels of a mixer also feature a stereo input, which means 2 jack inputs controlled by the same fader; this way is also possible to connect both left and right keyboard channels to our stereo channel.

Obviously if we don't have d.i. instruments, we can still treat the last 2 channels of our mixer as all the others, to control microphoned instruments.

Group channel tracks and Fx tracks: on some mixer there are faders (in this example we're talking about the 2 blue ones at the bottom right of the mixer) that can control a full group of other tracks, so basically we can apply to a live environment the same rules we have seen for group tracks and fx tracks used on a DAW, when mixing (Click here for a dedicated article).
If the mixer has this feature (like this one on the photo, which has 2 group tracks), each one of the 12 channels will have 3 selections, near each fader: Mix, Group1 and Group2.
If we choose Mix, this channel will be routed straight to the stereo mix, so its final volume will be only decided by the Master Fader.
If we choose a Group instead (for example we can send all vocals on Group1 and all guitars on Group2), we will be able to control all the vocal tracks submix with the "Group1" fader and all guitars submix with the "Group2" fader, and obviously both these groups are controlled by the Master Fader, anyway.
Fx Track: this track it's represented in the mixer on our example by the only white fader.
On this track we can choose an effect or a combination of effects (for example Reverb + Delay), and decide the amount of effect to apply, then on the single channels we can choose the amount of signal from the single tracks that should be fed to the Fx track, and with the Fx track fader we can decide how much effected signal should be send to the Master Fader, or to a Group Track, and to the various Aux Sends.

For example I can choose to set the fx control on the vocal track on 5/10. This means that half of the dry signal of the vocal track will be sent into the fx track, and will be effected; then with the fader control of the fx track, I can decide the level of this effected part of the signal to send to the stereo mix, or to a group, or to an auxiliary send.



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HOW TO MIX A LIVE BAND (a guide for dummies) 2/4

Hello and welcome to this week's article!
Today we proceed with part 2 of our 5 parts article about how to mix a live band.
After we have set up everything on our stage properly, we should have a channel box with 12 numbered xrl jacks, that we should connect to our mixer respecting the numeric order.

Here is my channel list, but you can modify it according to your own taste:

1   kick drum
2   snare drum
3   tom 1
4   tom 2
5   floor tom
6   lead vocals
7   backing vocals
8   guitar 1
9   guitar 1
10 bass guitar 
11 keyboards / d.i. 1
12 keyboards / d.i. 2

If we want to make sure we will find immediately the channel we're looking for we can use also a strip of writable tape to put below all channels, so we can write under each channel what instrument is plugged.
Each channel of the mixer looks the same, and a typical single channel strip it's the one present in the classic Mackie mixers, and it is depicted on the top of this article.
Let's analyze it, starting from top:

- we got the Xlr and normal jack input

- then we have a high pass filter button which takes out everything below 75hz (which is useful, so
we can leave the lowest frequency area only to certain instruments, such as bass and kick drum).

- a Gain knob (that sometimes is called Trim)

- two Aux Controls (which control two stage monitors, and decide how much sound from this channel will be played on monitor 1, which is connected to the Aux 1 out, and to the monitor 2, connected to the Aux 2. On the stage plan of the first part of our article there were 3 monitors, so the mixer must have 3 Auxiliary outs and 3 Auxiliary knobs).

- an Eq section, to control high, mids and lows, which is good to correct some unbalancing in our tone or to make some instrument to pop out more. Often this section is used to tame the lows and make the sound more intelligible.

- an FX knob (that on the photo of the channel strip is absent but that is present on many mixers and that controls how much effect, like delay or reverb, will be sent to the channel)

- a Pan knob, which lets us decide the position of the instrument into the soundstage. Keep in mind that a live environment is different from a studio one, so apply less extreme settings than you would on a record!

- a Mute and a Solo control, to mute the track or to hear it alone.

- the Volume fader, to control the channel's overall volume.

Ok, now we have all jacks plugged in the right channels, and we know all channel controls.
What do we have to do?

We must decide the gain sensibility. With the gain control we decide how high is the level of the incoming signal, then we can apply all the variations we want on each channel (eq, pan etc..) and finally with the volume fader we will increase or decrease the overall channel level to fit that instrument better in the mix.

To set the gain sensibility we must use the Gain Setting Procedure for each channel.
This procedure will make us set the gain knob (the one located on the top of the channel strip) in order to let out mixer "hear" correctly the incoming signal. If we have a PFL BUTTON (pre fader listen) we should turn it on for the gain setting procedure, in order to hear only the gain control, bypassing the volume fader. When you press that button you will see the level of that channel's input on your mixer's meter (which are the green, yellow and red leds located on the right in the mixer).
What we want is to use the gain knob to find a position that lets us avoid the level to hit peak or to be too low: the level of the instrument should be halfway, with occasional peaks in the yellow area. Press only one PFL button at a time, otherwise the meter will show the combination of both channels and you won't know which one is too high or too low.

Once we have done this procedure for all channels, the Gain Setting Procedure is over and we should have each channel's gain set to the optimal level.
Now we can proceed adjusting the Eq, the Pan (and if we can, the Fx, because vocals usually could use some reverb) and the volume of each channel to fit them better in the mix, until we have a stable mix in which each instrument can be heard decently.


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HOW TO MIX A LIVE BAND (a guide for dummies) 1/4

Hello and welcome to this week's article!
This is the first part of a tutorial in which we will analyze the bases to mix a live rock/metal band, on a small venue (200-300 peope audience): basically the average small live gig.

The idea of this tutorial came into my mind watching so many gigs in which the bands rented the p.a. system (public address system, which is the system composed by a mixer, microphones, loudspeakers and so on) and tried to set it up without a clue of what they were doing, and not wanting to hire a live sound engineer for the night.
The result is usually a mess: the band doesn't hear itself and plays bad, the audience doesn't understand a thing and eventually everyone gets scorned.

Please note that this is not a super comprehensive guide, and the most expert live engineers will notice something that may diverge from their workflow, but I just wanted to give the bases in the easiest and most understandable way possible, to get a decent sound live.

Ok, let's start with a description of the average cheap p.a. that can be rented and that we will use for our example.

- a cheap 12 channels mixer
- 2 speakers for the audience
- 3 monitors for the stage
- 2 microphones for the vocals
- 5 microphones for the drumset
- 2 microphones for the guitars
- at least a dozen of xrl cables
- a couple of d.i.
- a channel box to gather together all the jacks from the various instruments on the stage to the mixer below.

So in this first part of the guide we will talk about SETTING UP THE STAGE.
Obviously microphoning a live stage is a simplified version of microphoning instruments for the studio (for example a drumset or a guitar amp):
the first thing obviously is to have clear in mind how the disposition of the musicians and the instruments, amps, cabinets, microphones and monitors should be (and the image on the top of this article is a standard stage-plan, with the position of everything on the stage, and for this serie of articles we will use that as "standard disposition").
Of course to set up the stage we will have to position the drumset if possible on the center, moved to the back, the microphones in the part of the stage closest to the audience, and the other amplifiers (and the keyboard) on the sides.
The monitors (in this case 3), should be set in a position in which they can be heard from 2 or 3 musicians, so we can set them like depicted on the stage plan, maybe turned a little towards the singer, if needed. A third monitor should be used only for the drummer, and in our example we have put it on the side of the drumset.

Once we have decided where everything should be, it's time to plug in all the amplifiers, to check out if we have enough electric plugs for everything, and then we can start microphoning everything.
For the drums, we have 5 channels available, and we will use them for:
- kick
- snare
- tom 1
- tom 2
- floor tom
Notice that we are not microphoning the cymbals, because in the small venues they are clearly audible even without microphoning, the bleed on the other drum microphones is sufficient.
Often drum microphones comes in a suitcase with clips to fix them right into the drum shells.

Now we will use 2 microphones for the 2 vocalists, then 2 microphones for the guitar amps.
To find the right position of the microphone towards the speaker, check out our guitar microphoning guide.
For the bass, often we won't have to microphone the amplifier, because many bass amps has a "mixer out" or "d.i." out, which is made to go straight to the mixer, so we will connect it from that out to our channel box.

Finally, the last two channels are reserved to the d.i. boxes, on which we can connect a keyboard (with the left and right outputs), or a computer, or an mp3 player with some pre-recorded base, or some other instrument that we need to amplify, or if we don't need them, we can just leave them alone.

We should write down exactly what xlr jack goes into the channel box (for example channel 1: kick drum, channel 2: snare drum, and so on), so that we will have the perfect control on everything when mixing.


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