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Saturday, December 28, 2013

HOW TO MIX ROCK / METAL DRUMS (a guide for dummies) PART 4/4



CLICK HERE TO READ PART 1/4

CLICK HERE TO READ PART 2/4


CLICK HERE TO READ PART 3/4


Hello and welcome to this week's article! We have finally approached the end of the year and to the end of this long tutorial on the many ways to mix a rock / metal drumset!
The snare is the focus of the album, along with vocals, and the equilibrium among vocals, snare and kick it's the axis of the mix, the one that sets the mood of the song (higher kick and snare and lower vocals for death metal, higher vocals and lower drums for soft rock): long story short, we need to get the volume and the tone RIGHT, otherwise the whole album will sound cheesy.

If we have just the snare top track, we can start by using a gate, to take out at least part of the cymbal bleed (not too strong, or we'll take away also some of the good harmonics of the snare), then a hi pass filter up to around 70hz and finally a low pass one down to 12khz, to cut away some of the left cymbal bleed.
Now we need to take away the low-mid "mud", which lies among 200 and 500hz: just find the right frequences and take them slighly down, without taking away too much body from the sound (you can use a frequency analyzer to locate exactly where most of the energy lies).
Now if the snare needs some low-end boost we can lift with a medium-width boost the area around 125hz, but beware because we risk to end up by adding again the "mud" we have just taken out.
The last and trickiest thing to do to enhance the tone with an equalizer it's to boost the highs: if we feel that the snare sounds too dark we can try boosting somewhere between 2'000 and 6'000hz: this is the most delicate moment of all, because now we are defining the colour of the tone that will hit the ear of the listener first, so we must not overdo and be aware that the higher the frequences that we're boosting are, the louder will be the cymbal bleed.

Once we have found the right tone for our snare top track we can compress it at a ratio up to 8:1 (we must choose the ratio according to the dynamic range of the track: if we have many low energy snare parts like press rolls, flams etc, we need more compression or they will disappear in the mix, while if there are not and the strenght of the hits is consistent, we can lower the ratio).
The attack should be around 10ms, and as always we should set the release at a level that the compression can go back to zero or almost, between a hit and the other.
If we feel that by compressing we have lost part of the transient we can use a Transient Shaper to bring back some of the snap of the snare, usually avoiding cymbal bleed, but don't overdo because too much transient enhancement can make the snare too much "in your face" and unpleasant: we must always look for that euphonic "sweet spot" :)

If we need to transform the snare sound too much to get a decent tone, then we have a problem: an excessively processed snare it's a snare that sounds fake, therefore it would be a good idea to record it again. If we can't record again and we need a way to add some punch on the snare we can mix one or more samples with the original sound, and if we're good, we will find a sample that compensates the lack of frequences of our original sound.

Another way is to Double the snare track: we can process the first one like explained above, and use the second one just to add the "snap". We can gate the sound until we have only the snare snap, then we compress it very heavily, use some frequency exciter and eq to make the "snap" very strong, similar to a fingersnap, and then we can mix this sound with the regular snare to get the same effect we would obtain by using a sample, with the difference that this way the sound comes all from our drumset.

Finally, if we have recorded the snare with 2 microphones, top and bottom, we're gonna have a second track recorded pointing to the snare wires in order to give some sizzle to complement the top snare tone. This track can be also very useful to add some body to the top snare tone: instead of boosting around 125hz on the top track, we can boost the same area in the snare bottom track and see if it sounds better. About the hi pass filter, we can take out a bit more, even up to 90hz if we want, the important thing is to take a look at the phase: we must try inverting the phase to see the version that produces less frequency canceling, and use that one.

On the compression side, if we use more than one snare track (top + sample, top + bottom, top + top copy), my suggestion is to route them into a group track and compress that one, with the same settings suggested for the snare top.
If we need it, we can also set a clipper at the end of the snare chain: this is sometimes useful because, during the mastering phase, the limiting traditionally takes out part of the snare transient, but if we use a clipper (with very soft settings) we can recover part of that lost transient (it's not easy to explain how a clipped sound helps recovering its transient while it should actually work exactly at the opposite, but it works: the software recreates part of the transient).
It's interesting to notice that unlike for other instruments (e.g. hi gain rhythm guitar), in which the compressor should be as transparent as possible and the moment you notice it, it means that is damaging your sound, compression for the snare is a real tone shaping tool, and if used without exceeding it can bring up the wires of the snare bottom, and enhance the sound with an explosivity that cannot be obtained otherwise.

About the Reverb: if our drumset sounds a bit dry (or we like a big '80s style snare) we should create an fx track with the best reverb that we have and set it in order to make it sound similar to a studio room.
Then we must equalize it with a high pass filter that takes out everything up to around 200hz, and send it to our snare track (or group), to our tom group and to our cymbal group track, from there we can decide the amount of effect sent to each track according to our taste.


I hope that this long drum mixing tutorial was helpful, and by the way the song is mixed and mastered by me for my band, Strider. Contact us if you want a copy of our latest Ep "The Black Lotus", also to support my work with this blog!

Happy New Year from Guitar Nerding Blog!


CLICK HERE TO READ PART 1/4

CLICK HERE TO READ PART 2/4


CLICK HERE TO READ PART 3/4


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Saturday, December 21, 2013

INTERVIEW: MATT BACKER



Matt Backer is an American guitarist and songwriter, which has played with some of the best bands and single musicians in the world, including Joe Cocker, Sinead O'Connor, Cher, Elton John, Alice Cooper, Bob Geldof and many others.
He also composes music for ads and soundracks, including "Buffy the Vampire Slayer". 
Here's our interview! 

GuitarNerdingBlog: Hello Matt and welcome to Atoragon's Guitar Nerding Blog!

Matt Backer: Hi Atoragon!

GNB: Tell us about your career. By visiting your website (http://www.mattbacker.com ) it appears that you've already had an impressive career, and have played with a lot of stars, included Alice Cooper (which is one of my all-times favourite artists), Elton John, Sinead o'Connor, Joe Cocker and many more.
Which are the ones that you consider your career highlights? Which are the artists that influenced you the most? Is there still some collaboration that you'd wish to do?

MB: I've enjoyed working with them all in different ways - you'll get different inspiration from Tony Hadley and Martina Topley Bird and ABC, but all good. I'd love to collaborate
with Bowie!

GNB: I've heard that you are a guitar collector! Tell us about your love for this instrument and about your favourite models!

MB: I'm in the studio playing my 50's Les Paul right now. I have a 1963 Stratocaster that I love, a 1933 National, a 1946 Gibson LG 1 and many others.

GNB: What do you think about the state of the music business? What are your thoughts about today's underground and mainstream music scene? 

MB: There will always be good independent music. I think the mainstream business is struggling.

GNB: What do you think about the digital music distribution? And what about the file sharing? How do you think the music business will evolve in the future?

MB: Digital is the future, but if we don't find a way to reward creators, there will be no future of the music business.

GNB: Let's talk about live music! Which have been the best gigs you have ever played? Do you consider yourself more a live musician or a studio one?
MB: There have been many. From Perugia at midnight with Sarah Jane Morris to Primo Maggio with Julian Lennon, Singapore Grand Prix with Banarama... all good. Live and studio feed each other.

GNB: Tell us some funny story: which one has been your best/funniest experience as a musician? And your worst one?

MB: One and the same. The time when I was joking with the singer, Suzanne Rhatigan by playing the guitar behind my head. She pulled my trousers down... And took my
underwear as well!


GNB: Since many readers or our blog are interested mainly in the technical side of the guitar world, can you tell us your studio and live equipment? Can you tell us about the recordings of your latest album?

MB: It varies enormously. As airlines have made it more difficult to travel with equipment, I've had to scale things down. When I flew to Key West the other week, I threw a Boiling Point and a T.RexReptile 2 in my suitcase because I knew the lovely Darryl Brooke at The Grateful Guitar was going to lend me his beautiful 1957 Telecaster. The cost of flying my GigRig pedalboard would be prohibitive, so I can only use that when things are being trucked. In those situations, I like to use
two amplifiers, particularly if there is 80's splang involved. I use an old Boss GT 5 - I have 4 of them with specifically whooshy 80s-tastic sounds programmed in. 
I normally have an early 90's Fender Vibroverb and late 60's Vox AC30 for those gigs - they warm up the digital side of things. 
I have a Fender Master Built Custom Shop Strat which lives in the truck (with a Flying V as a spare)

The nature of the gig determine the nature of the gear. 
I used my 50s Les Paul through a TwoRock amp I have under the stairs (like Harry Potter, only different) the other night. 
I used a 1950s Maestro amp (with a 4x8 cabinet) on stage and in the studio with Rumer. 
Most of that stuff was done with the 1959 Gretsch Jet Firebird or 1963 Fender Strat that are pictured on the front and back cover of "Idle Hands" I used the Strat and my 1933 National Steel on Laura Mvula's album.

Most of "Idle Hands" was recorded in London. 
We had a variety of amps and guitars - my Matchless Chieftain, Ian Shaw's Blackstar Artisan and 1970s Marshall Combo, my 1949 Fender Deluxe. 
Ian's Gretsch Electromatic got used a lot, as did the old Strat and the 1963 SG you can see in the "Man Who Stole My Life" video. 
Acoustics include Taylor 6 and 12 strings, my 1957 Martin 0017 and Ian's Guild whatever it is.
"I've Fallen" was recorded in LA. 
I had my Bastardcaster, and Gibson kindly lent me an arsenal of acoustic and electric guitars. 
Phil Jameson lent me a Matchless combo - I can't remember which model. 
"All That You've Wanted" was recorded in Julian's studio in France using...gasp...a Line 6 guitar through a Pod! Just goes to show. 
Ian, Julian, Grant Ransom, Julian Simmons and Tom Weir all make me sound good.

GNB: Is there any advice that you'd like to tell to our fellow guitar players?

MB: Play what you feel.

GNB: What does the lyrics of your songs talk about? Do you think that on a song it's most important the lyrical side or the musical one?

MB: I like both live and studio, one serves the other.

GNB: The interview is over! Tell us about your latest album, projects and tours! 
Thank you very much and we hope to see you soon live!

MB: I've just been nominated in four catagories by the Independent Music Network.
Favourite Single for Let's Art, Favourite Video for Let's Art, Favourite Male Artist and Favourite International Artist. 
Awards will be announced the second week of January 2014.

I contributed to Everything Changes by Julian Lennon. I spent eight years as Julian's band leader and he and I are good friends. I also contributed to his new video project "Through the Picture Window".

I'm finishing a new blues record with my producer Ian Shaw at Warmfuzz Records in Key West Florida. Hopefully be putting the finishing touches on it January 2014.

I'll be touring in 2014 with ABC and doing solo shows in and around London. You can keep up with me via facebook at www.facebook.com/mattbackermusic twitter at @mattbackerworld,
reverbnation at reverbnation.com/mattbackerUK and my website is now being update very regularly at mattbacker.com
Thank you!

Saturday, December 14, 2013

HOW TO MIX ROCK / METAL DRUMS (a guide for dummies) PART 3/4



CLICK HERE TO READ PART 1/4

CLICK HERE TO READ PART 2/4


CLICK HERE TO READ PART 4/4


Once we have analyzed how to treat the room, cymbals and toms, it's time to talk about the kick.

Let's start by saying that there are different ways to work a kick effectively, according on if we want to preserve the acoustic sound or, at least, part of it:

Assuming that we have tracked a good sound, we must first do a high pass filter, cutting everything below about 35hz, then we can scoop an area that may vary from 150hz to 400hz: this is where the "mud" lies, but beware because if we take out too much energy from here, the kick will lose its body, especially on the cheapest speakers.
Once we have tamed the frequences where most of the energy lies, we can boost if we need some more "thump"  in the 50hz-80hz area, and some high end between 2'000hz and 10'000hz, depending on the type of coloring that we want to give to our kick.
On the compression side we can compress at a ratio around 8:1, with a fast attack and a release speed that should be set according to the song's speed.
Just remember that a too hard compression can ruin the sound's transient, so if we need to apply so much compression that the sound becomes dull, it's better to stack 2 compressors with lower settings, and/or try to reconstruct the lost transient with a transient shaper.

If we have a sub kick microphone too, we can consider our kick sound splitted in 2 tracks,  one for the lower frequences, one (with a high pass filter at around 500hz) for the higher ones: this way we can compress the low end more and the higher end less (thus preserving the transient), and maybe do some sidechain compression with the bass, so that the bass sound is slighly compressed when the sub kick track kicks in, which may be crucial when dealing with death metal speeds, to preserve clarity and separation.

Even if we are tracking our drums just with one microphone, we can always double the track and do the same as if we had 2 microphones: the result will be less complete, but always more flexible than dealing with one single track.

Finally, we can obviously use a sample: usually pre-processed samples tends to be oriented to the high frequences, so when we blend them with our acoustic sound, it's often a good idea to treat the acoustic track as the "sub kick one", and use it to emphasize the lower region.


CLICK HERE TO READ PART 1/4

CLICK HERE TO READ PART 2/4


CLICK HERE TO READ PART 4/4

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Saturday, December 7, 2013

MIX ALL SONGS IN THE SAME PROJECT (and track routing)



Hello and welcome to this week's article!
Today's post is an addition to our Project Preparation article: we talk about an alternative way to mix an album, that is possible only with the latest computers, now that ram and cpu are commonly enough to manage a high amount of tracks and plugins.

Instead of creating a single project for each song, we can create a big project in which record (or import) the takes for all songs.
Obviously the amount of tracks will be much higher (I have finished mixing a 10 songs / 101 tracks project just a few weeks ago), but this will also give us the advantage of setting levels that will affect the whole album and give to all songs a homogeneus feel.

First off we must se the Tempo Track, since our songs will probably not be all on the same click (here is a dedicated article about the tempo track).

Dealing with so many tracks forces us to organize the workflow in the most rational way possible, which means that the channel routing becomes crucial: we'll need to reduce as much as possible the processing of the single track, and focus in dividing the tracks in group channels, e.g. we have 8 rhythm guitar tracks, we can route all of them to the same "rhythm guitar" stereo channel, same for the 6 vocal tracks, and so on, so that in the single tracks we only set the volume level, the panning and load in the insert (only if strictly needed) the plugins that are specific for that single track (e.g. a "lo fi" eq effect).
Sub groups (like for example "rhythm guitars" and "lead guitars") can also be routed to another stereo group ("all guitars") for further common processing, like adding a compressor: we could surely use the compressor in the 2 sub group track inserts, but if we use it only in the "all guitars" group track, we will run just one instance of the plugin instead of 2 with the same result, saving some cpu.

Now it's time to think about the effects: we must rely as much as we can on the Fx Tracks (click here for a dedicated article), which are crucial to save cpu and ram. The Reverbs, in facts, especially the convolution ones, are some of the single most cpu hungry processors around.
We should find a few good effects (delay, reverb) and use them, when needed, on the fx sends of all the tracks of our project that needs some, instead of opening a new instance of each effect in the insert of the single tracks.

Last, once all levels of the tracks are stable, the editing phase and the sound sculpting one are over, we will find that we're still gonna have to do some adjustment in terms of volumes in the single songs, for example on a ballad the drums should probably be less violent than on an uptempo, therefore we will need to Automate the tracks in order to fit them better in the mood of each song.

Once we have done we'll be probably ready for the Mastering Phase, with a bunch of coherent songs and having spent less time than mixing each one on a separate project.


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Saturday, November 30, 2013

HOW TO MIX ROCK / METAL DRUMS (a guide for dummies) PART 2/4



CLICK HERE TO READ PART 1/4

CLICK HERE TO READ PART 3/4


CLICK HERE TO READ PART 4/4


Hello and welcome to this week's article! 
Today we proceed with the second part of our tutorial about how to mix a rock drumset.
First off we must decide our drum routing; starting from the 8 tracks acquired in Part 1 of this tutorial, I would do in the following way:

1 Stereo Group track for all toms
1 Stereo Group track for all cymbals
                +
1 Mono Group track if we have both snare top and bottom microphones
1 Mono Group track if we have more than one kick track (e.g. beater microphone and subkick)

Since we have already talked about how to mix cymbals (Click Here for the dedicated article), today we're going to focus on Toms
Let's assume we have a number of 2 or more acoustic toms (included the floor tom) on our group track: we can do most of processing directly on the group track, instead of process them individually.
First off we must Pan the toms, using the panning tool on the single tracks: I like to pan toms in order to make them be heard from the drummer's perspective, so usually with the floor tom on the right, and the other toms somewhere in the left and right soundstage, but not in the middle.
Then we can switch to the group track and apply some stereo processing (make sure that all the processors in the insert are stereo, or part of the sound coming from the panned tracks will not be affected!

Gating: this process is optional; it determinates how much "room bleed" you want to leave in these microphones. There will be some crash cymbal bleed and other sounds, and if we leave them all, the overall drum sound will be more natural, but it will also sound more "garage rock", more "alternative", and less tight. For thrash metal and other extreme genres, it's suggested to use a gate that takes out everything and leaves only the tom sound. 
Eq: the first thing to do is to apply a high pass filter starting anywhere from 40hz to 90hz, according to taste, then we must locate (using a frequency analyzer, click here for a dedicated article) where most of the energy is, and lower it. 
As for the Kick, acoustic toms are often full of low-midrange, that doesn't add too much to the sound but takes away a lot of headroom, so I'd suggest to find where the resonance lies and lower it, and it could be anywhere from 100hz to 800hz. 
Now that the low-mids are tamed we can also, if needed, raise the attack of the toms to make them cut more through the mix by raising some db in the 2 to 4khz area.

Compression: since toms can have a high dynamic excursion, it's important to keep them steady with a good compression: we can start from a 4:1 ratio and raise it, if needed, keeping 3 to 6db of compression (a good idea is to set a slow attack and a moderate release, to let some of the transient to pass before start compressing). If needed, because we still can't tame all the hits, we can also stack a mono compressor on the single tracks with a stereo one in the group track, or adding a Limiter after the compressor
Another alternative is to put a mono compressor in the single tom tracks, and a stereo multiband comp in the group track just to tame the peaks on the resonance area leaving unaffected the rest.
The idea is to be able to get the toms at a level that makes them always audible without the risk of a hard hit that suddenly covers everything else.



Saturday, November 23, 2013

INTERVIEW: BOB MILLER


Bob Miller is an American Musician, singer and guitarist for the band Nimrod Wildfire, among many other projects; his influences are, according to his website Villamilla.com, electric blues and Stax soul, 50′s rock & roll and doo wop, good pop songwriters, 60′s soul, Motown, funk, New Orleans piano, zydeco, country, jazz and latin.


GuitarNerdingBlogHello Bob and welcome to Atoragon's Guitar Nerding Blog! Introduce yourself to our readers, tell us your story!

BobMiller: I saw the Beatles on the Ed Sullivan show in 1964 and knew what I wanted to do with my life. It took me another five years to get a real guitar, but within a year I had my first band together. I had to teach the bassist and other guitarist their parts, but we had a great drummer. I was lucky enough to live in London by that time, so I came of age in a very demanding but collegial atmosphere. It was the golden age of pub bands, and my last UK band played some great gigs. I moved back to the States to go to Berklee and found a very discouraging scene in Boston, so after a couple of years I got serious about a day job and wound up running my own software business. Interestingly enough, most of the people I knew at Berklee became programmers. At any rate, I sold that business about five years ago to devote myself to writing, recording and performing my own music. I think the world still needs music that says something and I think I've got something to say.


GNB: Tell us about your career. By visiting your websites ( http://www.nimrodwildfire.com/ and http://www.bluemuze.com ) it appears that you've already had an interesting career, taking part as both studio and live musician in different projects.
Which are the ones that you consider your career highlights? Which are the artists that influenced you the most? Is there still some collaboration that you'd wish to do?


BM: I did some session work in London and loved it. I drove to Rockfield Studios in Wales once, sitting on my amp in the back of a Ford Transit van. I thought I had really arrived at that point. Gigs at the Northern Counties were great. Lately I've played some big festivals in this area. Any time the sound is good, the band is good and the people are appreciative it's pure joy for me. Likewise working on my last CD was great - both working on rhythm tracks with the rest of the band or working out a solo or vocal in my home studio.

I have pictures of Duane Allman, Ry Cooder, Richard Thompson, Ralph Wash, Lowell George and Django Reinhardt on the wall in my studio. They're the guitarists that have most inspired me. As a songwriter, which is more and more how I see myself, I like Nick Lowe, Graham Parsons, Jackson Browne and John Hiatt. There are so many more though! I listen to a very wide range of music. I'm one of the few people I know who thinks that Chuck Berry was a brilliant lyricist.


GNB: I've heard that you are a guitar lover and that you build your own instuments! Tell us about your love for this instrument and about your favourite models, as well as your creations!
BM: To be honest, it's mostly that I'm demanding but cheap. I can't afford vintage instruments, and having watched a cymbal fall and cut a gouge in my guitar on a gig, I'd really worry about gigging with something irreplaceable. I find that between Warmoth, Seymour Duncan and Joe Barden I can put together a really great Fender style guitar for a lot less than it would cost to buy the real thing and then switch out half the parts. I build my own amps too, for the same reason. My favorite is a recreation of the 1964 Vibroverb. I took a good year to design and build but it's a really unique sound.

I suppose I'm not terribly romantic about guitars. They're tools, but I've always loved tools! I tinker with everything. I have a few Gibsons and I don't think any of them has the original pickups or electronics. I just want them to play well and sound good, and I really hate hum, so they're well grounded and shielded.


GNB: What do you think about the state of the music business? What are your thoughts about today's underground and mainstream music scene?

BM: When I was at Berklee in the late seventies the word was that the music business was nothing like it had been five years earlier and it's continued to get worse. I think the problem is that as a society we don't really value creativity. It's not just the music business - nobody gets paid for Ted Talks either. I think our values as a society are really upside down. I'm speaking mostly of the States, since that's where I live, but I think this has happened throughout the western world. I think you get what you pay for and I really worry about the culture we're creating.


GNB: What do you think about the digital music distribution? And what about the file sharing? How do you think the music business will evolve in the future?

BM: As I say, I think the problem is more societal than technological, but certainly as music is currently distributed and consumed it is hardly viable to live as a creative musician today. I believe, and I see constant evidence, that there is a hunger for the kind of passionate, genuine music that I fell in love with as a kid. I believe therefore that things will change in such a way that such music will be created and the creators rewarded. Right now I really have no idea how that will happen, but the reason I'm doing this now is that I want to be part of it.


GNB: Let's talk about live music! Which have been the best gigs you have ever played? Do you consider yourself more a live musician or a studio one?

BM: I love them both. The two gigs that come to mind are a very hot, sweaty night on a tiny stage in a packed club with a blues band, and a beautiful outdoor gig on a large stage. So the thing really is as I mentioned earlier - good sound, good players and an appreciative audience. Everything else is peripheral.



GNB: Tell us some funny story: which one has been your best/funniest experience as a musician? And your worst one?

BM: It's only funny looking back on it, but I was in one band where I'd have to go find the bassist and lead singer after every break. I would wander the streets around the gig looking for a car with a copious cloud of smoke inside and knock on the window. Programmers aren't easy to manage but I never had to do that in the software business.
The worst of it is putting myself out there and not getting a response. Sending out CD's to the wrong radio stations or nagging people for a gig only to have them offer dates I can't do. I've been in some pretty dicey musical situations but no matter how bad it gets, playing music is the good part.


GNB:  Since many readers or our blog are interested mainly in the tecnical side of the guitar world, can you tell us your studio and live equipment? Can you tell us about the recordings of your latest album?

BM: We did the rhythm tracks for Corporate Refugee in London, with Jim Kimberley and Matt Backer, who played those pub gigs with me so long ago. Then I came back and layered keys, vocals and guitar at my leisure. My studio here is digital - I use Sonar, Reaper and ProTools at various times. I get sick of the shortcomings of one and then switch to the next for a while. Guitar never goes direct - I go through an amp and mic it. I'm a fan of Oktava microphones, modified by Mike Joly. Other than that it's pretty standard stuff - Shure, AKG.
My daily amp is an Allen Accomplice and there's not much you can't do with it. It's based on the blackface Deluxe Reverb but you can put 6L6 tubes in it for a little more volume and clarity. You can dial the tone stack in or out and get a nice tweed sound out of it. And it's not too heavy. In London we used to lug around a Hammond B3 and a Leslie, in addition to the Marshall 4x12's. No more!

I have a big Pedaltrain board and a small Pedaltrain board. For distortion I use Keeley OCD's, TS808 Tube Screamers, and a Zen Drive. I have a Keeley compressor and an MXR, and I use the MXR envelope filter instead of a pedal because I generally have enough to worry about without bringing my foot into the equation. I control them all with a microprocessor based loop controller that I built myself. It's kind of like the Carl Martin Octa-Switch but much smaller. I had the Octa-Switch and it worked great, but there wasn't much room for the pedals!

I also use various preamp pedals just to compensate for the different output and tone of different guitars. I don't want to have to fiddle with my amp when I switch, so everything is preset. Likewise, especially since I'm the lead singer as well, everything on the pedals is preset and I just have four loops to choose from, with a couple of effects that will be on or off depending on the song. I make it as simple as I can and I still mess up from time to time.

GNB: Is there any advice that you'd like to tell to our fellow guitar players?

BM: This probably isn't what you want to hear, but I always think of a story I heard about Eric Clapton plugging a borrowed guitar into a cheap transistor amplifier. He plunked a few notes, turned a couple of knobs, and sounded just like Eric Clapton. The passion in your heart and the feel in your fingers is 98 percent of it. I love all the gear, but you've got to keep it in perspective.

GNB: What does the lyrics of your songs talk about? Do you think that on a song it's most important the lyrical side or the musical one?

I learned a lot more from Muddy Waters than my school teachers when I was growing up. I've had a pretty interesting life, had some great times and made some big mistakes. That's what my songs are about, and I'm hoping somebody will learn from them as I did from Muddy Waters. If you're putting lyrics to music then they're really inseparable. It's like asking whether treble is more important than bass. I try to create a compelling whole.

GNB: The interview is over! Tell us about your latest album, projects and tours! Thank you very much and we hope to see you soon live!

BM: I'm writing the songs for the next Nimrod Wildfire CD at the moment. Gigs are infrequent in this area, but we play out as much as we can. I've spent a lot of very happy time in Italy and I would love for you to see me live too!

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Saturday, November 16, 2013

HOW TO MIX ROCK / METAL DRUMS (a guide for dummies) PART 1/4



Hello and welcome to this week's article!
Today we're going to talk about one of the most interesting topics of all, and one of the most complex: how to mix a drumset.
First off let's say that there isn't just one way of mixing a drumset, the variables are endless and everything changes according to the genere, the drum parts tuning, the number of microphones, and the choice of using samples or not.
If we want to use a drum sampler, in facts, we can take the drummer's playing via triggers, and then use a drum sampler to assign to each trigger a virtual drum part. Today many bands record their albums like this: the drum parts are played and the sounds are replaced by midi drum samples, and only the cymbals, the thing that sounds more "fake" when not microphoned, are tracked traditionally.

Speaking of acoustic drum tracks, instead, let's assume that we have an 8ins usb interface, and 8 microphones (click here for a dedicated article on how to mic a drumset). I would track the drumset in one of the following ways:

acoustic rock drumset

1 microphone for the kick
2 microphone for the snare (one top, one bottom)
2 overhead microphones for cymbals.
1 microphone for the hi hat
2 microphones for the 2 toms.

acoustic rock drumset alternative

1 microphone for the kick
1 microphone for the snare top
2 overhead microphones for cymbals.
1 microphone for the hi hat
2 microphones for the 2 toms
1 microphone for the room

acoustic metal drumset

1 microphone for the kick
1 microphone for the snare top
2 overhead microphones for cymbals.
1 microphone for the hi hat
3 microphones for the 3 toms

Obviously if we have more Ins in our audio interface we could really use some more microphones, like a room mike, a ride cymbal mike, a subkick or more tom microphones, if the drumset is bigger (Click here for a dedicated article on how to mic a drumset): I have decided to go just with 8 tracks because that's the standard amount of inputs most of usb audio interfaces have.
If we are planning of mixing samples with the acoustic sound, we can also use just one microphone for the snare, and use the spare interface In to use a room mike, a large condenser one. The room microphone is needed more if we are mixing a rock/hard rock song, and less if we are mixing an extreme metal tune.
If we want to mix a metal song and we are planning to blend some sample with the acoustic sound we can also just switch the snare bottom mic to an eventual third tom.

Obviously we can also decide to use microphones just for cymbals and use triggers for all the other drum parts, so that the when the drummer hits the skin the trigger sends a midi impulse to the computer, and the impulse is replaced by a drum sampler.

Since we have already covered the topic about how to mix cymbals (click here to read the article), here we will focus on the whole drumset and on the remaining drum parts: Snare, Kick and Toms.
Assuming that we have a set of 8 nice and clean tracks, recorded at the right levels, with the drum parts tuned to perfection, a drummer that played well, and all the microphones in the right place, we can start editing, until we feel that the song is precise enough to move on with the mix.

Before star sculpting the sound of the single drum parts we must make a background choice: are we using a mix buss compression? In that case we're going to keep it in mind when compressing the single elements, since the two stages of compression will stack (click here for an article about serial compression). The situation will be even more complex if we decide to apply a specific compression on the single elements, then a compression on the whole drum buss, and then a whole mix buss compression. In this case, we must lower each compressor settings accordingly, in order to balance between them and have a final result that doesn't sound over squashed.

If we are dealing with samples, since samples are 99% of the times already half-processed or completely processed, I would probably go with a soft drum buss compression (and only if the drumset needs more punch), just to glue together the elements and not to squeeze too much the transients: the ratio should be very low, like 2:1, the release should be around 100ms (it depends on how fast our track is, what matters is that that the release "matches" the time of our drums), and the attack should be slow as well (e.g. 30ms, otherwise we risk to lose the "snap" of the snare), in order to let the transient to pass and then to add tightness.
If we are not using a Room Microphone we can recreate it with Parallel Compression: we must route all drum tracks to a separate buss, and compress it very heavily (and, if we need more room, we can also add a reverb), now with the fader of this track, we can decide the amount body to add to the whole drumset.
If samples are completely unprocessed, we can also process them as if they were recorded acoustic, but obviously we won't need a Gate.

If we are dealing with acoustic drums, instead, the need of a drum buss compression depends on the amount of compression we're applying on the single drum parts (which could be enough), and if we're using a room microphone. The room microphone in facts can be used to add some parallel compression to our drumset: we can compress it very heavily (this will add a lot of body to our drum sound), and then adjust the volume of this track just to decide the "amount of body" we want to add to the whole drumset, and this is a great alternative to drum buss compression, that doesn't touch the transients of the single drum parts.

Additional awesomeness: instead of using a Buss Compressor on the drum buss (maybe because we have already used strong compressors on the individual drum parts), we can also use a Virtual Console Emulator; it is a tool between a Compressor and a Saturation device, that can give to our sound just the character we need without over compressing it.
Another alternative is to use a Harmonic Exciter, to bring out some of the high frequences we couldn't obtain otherwise, but beware because this tool can breathe life into dull mixes, but it can also really screw up everything, so use it with caution.

CLICK HERE TO READ PART 2/4

CLICK HERE TO READ PART 3/4

CLICK HERE TO READ PART 4/4


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Saturday, November 9, 2013

SERIAL COMPRESSION (a guide for dummies)



Hello and welcome to this week's article!
Today we're going to talk about a common problem during mixing, specifically when mixing instruments with a very high dynamic range, like the Bass guitar and some drum parts.
The idea behind the Compression, as we have seen on the dedicated articles, is to make the sound "stable", to avoid it to move around in the mix balance: this is particularly important for the Bass, since its low-oriented sound generates a lot of energy, therefore it needs to be "tamed" with a particular strenght.
Compression, when set with a high ratio (e.g. 10:1) works almost like a Limiter: it attenuates the peaks that surpass a set threshold with a very strong reduction (in our example every db over the threshold is reduced to 0,1db).
A Strong Attenuation generates obviously a pretty drastic cut on the compressed wave, and it may damage the Transient (click here for a dedicated article) in a very bad way, for example if a Snare track is overcompressed, we will notice that the sound will lose its "snap" and will become very unpleasant.

So How do we tame an highly dynamic sound preserving the transient from being excessively cut?

We can try using a serie of two Compressors, one after the other (thus the term "Serial Compression"): the first one will just tame part of the peaks, moving the whole sound towards a narrower range, the second one will peel off the remaining peaks, and if we set both compressors for example at a ratio of 5:1 we will notice a perceivably better transient preservation rather than using a single, 10:1 ratio instance.
The result will be a Bass sound that will never move from where we set it, and this will allow us to raise its volume without worring.

About the Drumset, the whole situation is a bit trickier: every drum part needs a different compression, but what is important it's that usually all the drum tracks are routed to a drum bus, in which is often set a buss Compression (click here for a dedicated article), therefore it's important to balance the single track compressions with the buss one, in oder to not squeeze the sound too much.

Sometimes the tracks are even compressed on a global mix buss and then compressed again in the Mastering Phase, before getting Limited.
It's EXTREMELY important to have clear in mind all the stages of compression that will be applied to the single tracks, and lower the single compressor settings accordingly, or the final result will be very unpleasant.


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Saturday, November 2, 2013

The difference between Vst 1, 2, 3, and between 32bit and 64bit Plugins! a guide for dummies



Hello everyone and welcome to this week's article.
Today we're going to understand the difference between 32 and 64 bit Vst plugins.
First off let's say that the older computer generation features a 32 bit processor, a 32 bit OS and a 32 bit Daw, and this is the way it has been for the last (at least) ten years, and in many studios it's still today the standard.
A 32 bit Os and Daw can only handle less than 4gb of Ram, which means that if you add more memory slots, they will not be detected from the computer.
With the increasing of the computing power needed to run the latest plugins, 3 gb of RAM weren't enough anymore to handle big projects, so lately the market has seen the blooming of 64bit cpus, often with multiple cores, that can handle much more than 4gigabytes of Ram.
With a 64bit cpu we can run a 64bit Os and a 64 Bit Daw, and this chain will let us use all the Ram we have installed, with a lot of benefits in terms of speed and stability (actually a 64bit cpu has many more upsides, but we're musicians, not computer techs so it's not our concern :D).

If we can, it's very important today to have a 64bit processor and Os, in order to be able to use all the ram we can, that is one of the most important things needed in digital music production.

What if we have a 32 bit Daw and it doesn't see our 64bit Vst plugin, or we have a 32 bit plugin that is not being loaded on our 64 bit Daw?
There is some Bridge program, like Jbridge, that helps us in this transition phase, until we will all have only 64 bit software (some Daw already features a bundled bridge program, though).

The 32 and 64bit plugins topic is also a obviously connected to the Vst standard.
Vst plugins have evolved in terms of optimization from their first standard, to the version 2, up to the latest Vst 3 version, which is specifically optimized for the 64 bit computers, and offers more routing options, along with other upgrades that makes this standard more stable and cpu-friendly.

(the video on the top of this page is a song mixed and mastered by me for Subcortical Inertia, using 64 bit Vst plugins).

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Saturday, October 26, 2013

WAH WAH and the other ENVELOPE FILTERS! with Free Vst Plugins Inside!



Hello and welcome to this week's article!
Today we're going to talk about the Wah Wah, and more in general about the envelope filters!
Envelope filters are basically Equalization filters that can work both automatically, through fixed or dynamic time and amplitude parameters, or can be controlled manually, to bring in and out in real time certain frequences from a sound.
The Wah Wah, which is the most famous application of an envelope filter, is a foot-controlled filter that takes in or out certain treble frequences, while a note is sustained.
The name Wah Wah is onomatopoeic, which means that the device recreates the sound effect produced, in facts if we apply it to a trumpet or a distorted guitar, the result is similar to the modulation of vowels made by a human voice (in facts one of the most famous wah wah pedals is called "Cry Baby").

One of the musicians that contributed to make the Wah Wah pedal famous is without any doubt  Jimi Hendrix (impressive how this artist has been taken by us many times as an example for the use of effects: the truth is that after half a century, the guitar effect standards set by him has never really gotten old), with songs like "The Burning of the Midnight Lamp", or Guns'n Roses guitarist Slash, with classics like "Sweet Child o'Mine".
Another artist that has built his fame also around the sound of his wah is Kirk Hammett of Metallica.
There are many different types of Wah, that changes according to how many and which frequences are filtered, but the idea behind them is always the same.
Talking about the envelope filters in general, instead, we can see a massive use of them in todays electronic music, for example artists like Skrillex shapes their tone by filtering in and out certain frequences from their lead synths, in time with the song to emphasyze the groove.


Most of DAWs already features a bundled filter or wah effect, but if you want to try some cool vst downloadable for free, here's our suggestions:

- Samsara Cycle Audio Wahzi: an interesting autowah effect with many editable parameters.

- Mtg Wahwactor: a very simple wah emulator, Voice Controlled.

- Coyote Electronics WahGT: a wah emulator that can also be controlled with a midi pedal.

- Fretted synth WAHT: a wah designed for the guitar.

- GSI EFFECTIZER: a single vst that features all the modulation effects, included the wah.

- Admiral Quality Naive LPF: a midi controllable envelope filter.

- Topaz Dyna Filter II: an AutoWah and Dynamic Filter.



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Saturday, October 19, 2013

THE PERFECT BASS EFFECT CHAIN ORDER



Hello everyone!
Today we talk about an interesting bass related topic, that we have already seen for the guitar (Click Here for the perfect guitar effect chain order!).
The rules aren't totally different, actually they are very similar, but we need to adapt them to this other instrument's characteristics.

So, as for guitars, here is our bass chain by areas: BASS -> PRE GAIN AREA -> GAIN RELATED AREA -> POST GAIN AREA -> AMPLIFIER

- PRE GAIN AREA: Starting from the instrument itself, it's a good idea to set at the beginning of the chain the WAH or the othter envelope filters (such as the Electro Harmonix Bass Balls), then we can set an Octave, and, if we need it, a Compressor.

- GAIN RELATED AREA: in this area we can use any kind of Overdrive, Fuzz or Distortion, and in this case the Post Gain effects goes after this area, or we can use the Amplifier's Preamp section as Gain Related Area, and the Post Gain effects goes into the amp's FX Loop.

- POST GAIN AREA: in this area we can set all the post-gain processors, such as modulation effects (chorus, delay..) and last the Reverb.

Here is the effect order: BASS -> WAH/FILTER -> COMPRESSOR / LIMITER ->OCTAVE / SYNTH -> OVERDRIVE / DISTORTION / FUZZ -> CHORUSFLANGER / PHASER / TREMOLO -> DELAY / REVERB -> AMPLIFIER

As for the guitar, the EQUALIZATION can be set anywhere, from the top of the chain to right after the gain related area; the thing to keep in mind is that if we cut, we can go before the compressor, if we boost we should place the eq after the compressor, otherwise it would cut our boost.

Keep in mind that this is just one of the many chains usable, and it's made to understand the basic rules of bass effects, but we can obviously experiment modifying the effect order to see what happens, and often the results will be very cool and creative. 
Just keep an open mind and let us know if you find some interesting variation!

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Saturday, October 12, 2013

REVIEW: 1973 GIBSON SG STANDARD


Hello and welcome to this week's article!
Today we're going to review an interesting guitar, owned by the father of a friend of mine.
This guitar is one of the very few Gibson that featured a Bigsby bridge, the type of bridge that allowed the player to use the whammy bar whithout having to dig a hole in the guitar body, and this feature disappeared almost completely from Gibson guitars around the half of the '70s.
The body is extremely resonant, yet not too heavy, this may be also because it has been regularly played for 40 years and always kept in dried places, allowing the wood to dry out and become resonant at its best.

The pickups were also different from today SGs: they were still humbuckers but with a lower output (some sort of "p.a.f."), still very creamy and mid-rangey, but with more clarity and less bass than the contemporary models.
These same pickups, treated with tar (the way they used to make them a few decade ago), today are equipped only in some dedicated vintage reissue.




Tech Specs:

Body: Solid mahagony
Neck: Mahagony
Fretboard: Rosewood 22 frets / block-inlays
Pickups: 2x ori. Gibson "Pat. Pending-stamped" tar humbuckers
Electronics: 2x Volume, 2x Tone, 1x 3-way toggleswitch
Pickguard: Small ´60s style 5-ply pickguard (blk/wht/blk)
Tuners: 6x Grover metal
Bridge: Bigsby


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Saturday, October 5, 2013

REFERENCING and MIXING!



Hello and welcome to this week's article!
This article can be considered an extra part of our Ear training section, and explains practically how to learn something about a professional record's song and to translate it in our mix.
Let's picture this scenario: we have our mix project in our Daw, we add a new stereo track and import our favourite song, lowering its volume until it matches the one of our project.
Now we can investigate that song, asking it the typical questions that we have already seen on the second and last part of our ear training aricle:

"how much room is reserved for rhythm guitars? Where is located the snare? How deep the bass goes? how much room is taken from the vocals?" ...and so on, and we use the tools and the methods explained in those articles to find out how our favourite mix is composed.

Once we have clear in mind how the song is constructed, tonally speaking, we need to compare it with ours, keeping in mind that 60% of the tone shaping is in the recording, and that we cannot twist a sound into a completely different one without runining it: we can cut frequences doing relatively little damage to the nature of the sound, but when boosting, the more we boost, the more it will sound unnatural and digitally warped (as it actually happens).

Comparing our song with the referincing one we coud notice for example that our snare drum sound it's completely different from the imported one: that one for example can be high oriented, because the drummer had a small snare with a very high tuning, while ours is dark because the snare was huge and low tuned, so the sound will never be similar, and it will sit on a totally different place in the mix. We cannot completely transform it because it's gonna sound awful, so we got two options: replace our snare with a sample through a drum replacer, or compare our track with another one that has sounds more similar to ours. It has no sense to have for example a grunge song to mix, and compare it with a death metal one only because we like the snare: it will only confuse us and lead us to wrong choices.

What it's particularly interesting is to understand how the audio engineers have treated the 2 most problematic areas: the one in which the vocals and the snare drum lies, and the low end.
The first one it's the zone the human ear is most sensible to, so we can understand by listening to it how a mix is balanced: are the vocals at the same volume of the drums, as in the modern rock?
Are kick and snare louder than vocals (as in death metal)?
Or the opposite, as in pop music?
We can apply the same pholosophy to our mix, and it will make the difference.
This area is also interesting to study, to understand how the engineer has treated guitars: has he cut the guitar frequences here to leave room to vocals and snare, or did he let the guitar be prominent, sacrificing frequences of the other instruments on that area?
About the low end: where does the guitar stops and where does the bass pops out?
The lower the guitar tuning is, the more usually a guitar is low-frequency oriented and "eats" room to the bass.
Is the bass distorted? How much?
To the point of becoming just a complement of the guitar as it happens in many black or death metal albums?
Or at the opposite, it's just lows and it lies on a range reserved to it?
How loud the cymbals are? Usually the louder they are, the more the album acquires a "garage", realistic, alternative rock feeling, and the more it becomes "dirty".
In pop music and modern metal, cymbals are held very low in volume, while the indie bands likes them to be high and resonating, so, again, it's a style choice that we must understand and use to our advantage.

These are all lessons in sound balancing and eq choices that we can absolutely learn on our own from the masters, and apply them on our mixes, keeping in mind that in the end our mix has to sound well, to be euphonic and balanced with what we got, that is totally different from the original sounds tracked for the referencing songs.
We cannot be too stubborn in wanting a sound that we cannot obtain because the raw track is too different, and we should focus instead into making great music with what we got.
In the end, that is really all that matters.


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Saturday, September 28, 2013

BIT DEPTH and SAMPLE RATE! a guide for dummies



Hello and welcome to this week's article!
Today we continue with the technical stuff made easy, and we're going to find out what are and how to use the Bit Depth and the Sample rate.

First off: why do we need them?
Because we need to choose the right setting before starting to record our tracks: the different settings will affect the dimension of each recorded track, its characteristics and the cpu usage of the project itself, so we must find the right compromise between quality and usability, especially when it comes to large projects.
The types of bit depth and sample rate availabe is variable according to the type of Audio Interface, and can reach 24bit and 192khz (as a maximum), but if an audio cd, an mp3 or a Youtube video can read only files at 16bit and 44.1khz, why do we need to record at settings so different?
The idea behind the availability of those settings is that we can record at the maximum fidelity possible, and then Dither the signal down to the standard format, obtaining a result that can be better than recording straight in the cd format.

- Bit Depth: why does people record usually al 24 bit? Because Bits are all about headroom: with 24 bits we're allowed to record hotter signals without the risk of clipping, or to record normal signals with a much quieter "noise floor".
In facts, in the analog era, it was important to record with as much signal as possible before clipping, because that way there was much signal emerging from the noise floor, which was quite high; in the digital domain, especially with 24 bit interfaces, the noise floor is so low that we can perfectly record with conservative levels without risking anything.
Sometimes the DAW offers the opportunity to record at 32 bit, but since 99% of audio interfaces in the market can record at a maximum of 24 bit, this feature should not be used.

- Sample rate: to understand this parameter, think about the frames of a movie. If a movie is 24 frames per second it's good, if it's 60fps it's much more fluid.
Sample rate does a similar thing for the audio section: it controls how many snapshots of an audio wave are taken in a second.
The reality is that 44.1khz are enough to represent well basically any musical style, so unless we have some particular need that forces us to use different settings, 44.1khz it's perfect, and at the same time it doesn't weight too much both on our Cpu and hard drive.
Obviously if you want to record your album at higher sample rates feel free to experiment, and if you want, let us know the differences you've noticed!

Our suggestion for a project is to record at 24bit and 44.1khz, that way there will be enough headroom and a great noise to signal ratio, and at the same time the project will not be too heavy on the cpu and hard drive.

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Saturday, September 21, 2013

ASIO, BUFFER SIZE and LATENCY! A guide for dummies.



Hello and welcome to this week's article!
Today we're going to talk about a topic that comes before the beginning of a recording session: how to set up the drivers in our computer in order to be able to work without problems.

First off the Drivers: each Audio Interface has its own driver software, that "explains" to our computer how to communicate with it.
When it comes to using a Daw, the part of the interface driver that is important is the ASIO section (audio streaming input output): this is the protocol for audio transmission, created by Steinberg, that lets us record and play one or more channels at the same time, with a very low latency.
Once we have set the Asio driver of our interface in our Daw, the workstation will stream all the input and output audio data from our interface.
If we have no Audio Interface but we need to record something on a Daw, we can use a driver called Asio 4 All, keeping in mind that the integrated preamp of our motherboard is really cheap, and it's a good idea to buy a dedicated interface as soon as possible, if we want to produce music.

Now that we have all setup we can try to create or to open a project, and see if we can hear everything well or there are problems: we could experience some mistimed sound, or some random pop or crackle (notice that this errors are not present on the tracks, they are just playback artifacts generated randomly when the Cpu is under an excessive stress).
This happens because our computer, in order to let us hear properly all the tracks (with all the eventual real time processing) needs to buffer the audio prior to let us hear it: the smaller the buffer size is, the lower the latency will be;
making the buffer smaller comes at a price, however.
With a smaller buffer, there is less overhead for delays in processing, therefore the CPU will need to work harder to ensure that any delay is kept within the time allowed by the buffer, and the more the Cpu works and the smaller the buffer is, the more is likely to experience dropouts, pops and crackles.

What we need is a very good cCpu and a lot of Ram, in order to be able to keep the latency low without too many problems; we can start by choosing from the Interface driver a small buffer and as soon as we experience problems we can increase it of one step, until we find the buffer size that lets us work perfectly at the minimum latency possible.

Coming to the latency, we're obviously talking about the delay between when we play a note and when it is received from the computer, processed and sent back to us through the monitors (or the headphones).
When mixing, latency is not a major problem if all the tracks are playing in perfect synchronization among them, the problem is particularly tedious when recording: if we are recording a singer or a guitarist, the maximum monitoring latency tolerable is 10ms or less: with a higher latency it becomes almost impossible to play.
If the latency is unbearable the only solution is to reduce the buffer size, and if this brings pops and crackles, the only thing to do is to try to lighten the Cpu load by disabling all the unnecessary real time effects, or muting tracks, leaving on just the most essential ones.
The ideal is to use a smaller buffer when recording, and a larger one when mixing.

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