4. Techniques
-----------------------------------------------------------------------------
   Take your time
  ~~~~~~~~~~~~~~~~
   An essential point about producing a quality tune is the amount of
preparation you put in, before you even begin to start.  This is especially
important if you intend to produce in a style unfamiliar to you.  Take the
time to get good samples, and see how they could be made to fit together.
Listen to the style, you don't have to buy tons of new music, just see what
friends have lying around, and the radio can be a good source.  Play around
with various ideas in your tracker, you needn't save them.  Get hold of a few
MODs and see how they work.
   We're not talking about a few hours here, not even a few days.  It may
take a few weeks or even months before everything's ready.  But when it is
you should find that you're able to produce, fairly quickly, a quality piece.

   Spicing Up Your Percussion
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
   (Taken from CU Amiga May 1994 - Slightly edited to be more generic)

   Fat Beats
  ~~~~~~~~~~~
   There are a number of things you can do to add a bit of life to your
percussion.  One of the best ways to beef up a drum sample is to mix it with
another sample.  You've probably already experimented with this, mixing kick,
snare, and hi-hat samples, in order to fit your entire rhythm into one track.
However, to get a really kickin' sound, try mixing your percussion samples
with samples of tuned instruments.  For instance, mixing a really deep
analogue-type bass sound with a kick drum produces a really heavy, squelchy,
dance floor sound.  Similarly, try mixing snare and guitar sounds, for an
unusual and funky effect try adding Laser-type pulse sounds to 808 style
snares for an authentic Sheffield clunk and bleep sound.

   Echomania
  ~~~~~~~~~~~
   Another way to add a bit of life to a rhythm track made up of individual
samples is to echo the entire track.  This is a quick way of funking up your
percussion, and you'll find you can create a great track with only kick,
snare, and open hi-hat when you use echo in this way.

   Bring Out Your Dead
  ~~~~~~~~~~~~~~~~~~~~~
   You've probably got quite a collection of hackneyed breakbeats, which are
instantly recognisable, and therefore pretty much unusable.  One way round
this is to sample some more, but in theory at least, you always have to be
careful of the copyright laws when sampling other peoples material.
   You could always buy a sample-compilation CD, but most of these are a tad
expensive for the casual user.  On the other hand, it's quite possible to
breathe new life into a dead breakbeat.
   One method is to apply some sort of sound effect to the sample, preferably
in stereo.  Most sampling software nowadays has a range of effects built in
with which you can process you sample, but most of these produce fairly
unsubtle results when applied to percussion samples.
   So what's the alternative, if you can spare the memory and two tracks (a
stereo pair is what we're looking for here) is to use the tracker itself to
produce a real-time phasing effect.
   To do this, load the same breakbeat sample into two different sample
locations.  For best results, pick a breakbeat that stretches over two bars
(32 lines of a standard 64 line pattern).  Play the first instance of the
sample (at a reasonable rate!) on line 0 and line 32 of a 64 line pattern, on
one track.  Do the same thing on track 2, but this time with the second
version of the sample.  Now for the clever bit.
   Fine Tune the second version of the sample up or down one or two points.
Now when you play the pattern, you'll get a phasing effect, with the rhythms
moving in and out of the stereo field - great for trance techno type
extravaganzas.  If you're feeling particularly adventurous, try playing one
of the samples an octave down from the other.
   If you can't spare the memory or two tracks for the rhythm, you can get a
similar effect in mono as follows.  Load up the first and second breakbeat as
before, and resample or pitch shift the second by a few points, then mix them
together.  The effect is a lot less subtle than the stereo version, but can
be just as effective in the right circumstances.

   On A Ragga Tip
  ~~~~~~~~~~~~~~~~
   Another way to squeeze the last bit of life out of a dying rhythm is to
change the playing length and sample trigger positions from the normal start
of the bar.  This is a technique much favoured by breakbeat and jungle techno
groups like SL2 and The Prodigy, and works best at fairly fast BPMs. Play
your breakbeat on lines 0 and 32, and adjust the tempo so that the rhythms
trigger in time, with no glitches.  Now trigger the sample on the following
lines: 0,6,16,26,32,42,48 and 54.  When you play this back, you'll have a
rhythm track that sort of rolls around the beat - perfect for just adding a
baseline and calling it your finished song!
   For a brutal stereo version of this, try playing the same sample on a
different track (on the opposite stereo channel) on the following lines: 0,
10,16,22,32,38,48, and 58.  You might even go the whole hog and combine this
with the stereo phasing effect.

   The Zen of Tracking Advanced Tips and Tricks
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
   Indian Food for Thought
  ~~~~~~~~~~~~~~~~~~~~~~~~~
   You can get a very Indian-sounding "24-tone" scale in Impulse Tracker by
using this technique: (FT2 users will have to accomplish the same thing via
the "tone" setting)

   Load your sample twice.  Look at the second one, and write down the sample
rate.  Multiply that number by 1.0304 and put the result in the "playback
rate" field of the first sample.  Now you have a consonant tone in the second
sample and a semitone above that in the first.  By playing the second at C-5
then the first at C-5 then the second at C#5 then the first at C#5 (and so
on), you get a semi-tone chromatic, which is pretty weird.  If you're really
bold, you might get some cool Indian sounding stuff going out of it.  Good
luck tracking it, though.  It's a whole new set of musical theory.

   The Amiga Scene and You
  ~~~~~~~~~~~~~~~~~~~~~~~~~
   If you either release or listen to MODs (not XMs, ITs or S3Ms, etc), then
you're probably aware of the Amiga scene, which still uses the MOD format
today.  If so, hold this in mind: the Amiga plays music 1BPM faster than PCs.
For example, at speed '6' in a MOD, a PC is playing it at 120BPM (I would
assume, anyway), and an Amiga is playing it at 121BPM.

   What this means to you, the listener, is certain drum loops and riff
samples will sound off-kilter, rhythmically.  So be a little more forgiving
in such circumstances.  If you want to hear it as it was originally written
on the Amiga, put it in Fasttracker (or whatever your favourite tracker is),
save it as an XM (likewise with the favourites), and change all the tempos in
the song to their appropriate fine-tempos (BPM), plus one.
   (Remember to do the reverse if you're producing a MOD on the PC that'll
probably get played on an Amiga.  When the tune is finished convert all the
primary tempos to 1 less.  This BPM thing sometime gets more extreme too,
maybe 2 or 3 BPM out in certain circumstances.  ModPlug and various other
players overcome this BPM problem.  - Cools)

   There are also some other effects that don't convert well from Amiga to
PC, which are apparent in chip tunes.  For the best reproduction (though
still not perfect), look for a player called "Midas Player", since it handles
things a little better than most with MODs.

   Radix has a few things to add:

   Well; in ProTracker the EFx command is used a lot... it actually changes
the waveform in the sample (only in the beginning).  So in chip tunes, chip
sounds can get some kind of wave sequence sound, "weeeeeeeeoooong" that does
not work on any program on PC I have seen.  Arpeggio on PC is not that fun
either.  I don't know really, but chip sounds sound better on Amiga...
    Another thing is that PC with a GUS can sound really awful while playing
a high and a low tone of the same sample at once.  This is really lame.  Like
a C-3 and a C-7 (same sample) sound really out of tune.

   Very Cool Reverb
  ~~~~~~~~~~~~~~~~~~
   Sure, you have an echoed lead.  But do you have a reverberated lead?  This
sounds very cool indeed.
   Load the lead in your favourite sample editor (mine's Cool Edit), reverb
it however you like (I use a straight reverb, on the "last row seats"
setting), so that it's REAL deep.
   Now load the tracker.  Create the echo track as usual (copy the lead,
offset it by a few rows, and change the volume by less than 50% of the lead).
Much nicer, eh?

   If you want a reverb that's not-as-deep to use somewhere else, you can
widen it for the echo track, creating this weird echoed attack kind of thing,
like this (FT2 Format, 1 is the lead, 2 is the reverb):

          01  C-5 01 40 000    C-5 02 08 840
          02  --- -- -- 000    C-5 02 10 8A0
          03  --- -- -- 000    C-5 02 20 880
          04  --- -- -- 000    --- -- -- 000
          05  F-5 01 34 000    F-5 02 08 8C0
          06  --- -- -- 000    F-5 02 10 860
          07  --- -- -- 000    F-5 02 20 880
          08  D#5 01 3C 000    D#5 02 08 840
          09  --- -- -- 000    D#5 02 10 8A0
          0A  D-5 01 30 000    D-5 02 08 8C0

   Of course, you don't need to keep retriggering the note.  I just thought
it sounded cool with bouncing pan.  In any case, I think a reverberated lead
sounds even better than an echoed version.  Try it and see for yourself.

   Phased Leads
  ~~~~~~~~~~~~~~
   A very cool effect for writing leads, commonly used by advanced trackers,
is a phased synth string.  (In fact, it's almost hard to call this an
'advanced' trick).  You can find samples that work for this is a lot of
different places (any good 'sweep' string sample will do), but the way that
they're used is the important aspect...

   It's quite simple, really.  You just create an instrument with a volume
envelope typical of a lead.  Something with a sharp attack, a moderate length
sustain, and an exponentially quieter decay (my ANSI art is miserable, but
I'll try):

           .  <-- Full volume here
           |\______
          /        \
          |<-- 60%  \_
              volume  \__
               here      \____
                              \________
                                       \ <-- 10% volume here (or less), and a
                                              moderate (300ish) fadeout.

   The total length of the envelope should be about twice as long as the
average length of the note (i.e.: an average length of a quarter-note should
have an envelope that lasts about as long as a half-note).  Now, as you write
your lead, keep the notes in the same channel, and slide to them at a very
fast rate ('F', generally), like this:

          01  C-5 01 40 000   <-- This starts off the sweep
          02  --- -- -- 000
          03  --- -- -- 000
          04  --- -- -- 000
          05  F-5 01 34 3F0   <-- You slide to the note here
          06  --- -- -- 000
          07  --- -- -- 000
          08  D#5 01 3C 3F0   <-- And here...  See the effect?
          09  --- -- -- 000
          0A  D-5 01 30 3F0   <-- Etc.  Retrigger the note
                                  to 'start over' the phase.

   It's important, however, that you echo this lead in another channel, since
it will sound fairly flat otherwise.

   Sound & Sampling Explained
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
     By Rubz/Hertz

   As you probably know from physics, sound is essentially made up of waves
travelling through the air - sound is merely vibration caused by some object
or another.  Of course, that isn't entirely accurate, as sound can pass
through solids and liquids as well (in fact, the denser the medium, the
better the sound is conducted - that's why whales can communicate with each
other over distances of miles, because water is denser than air.)  The medium
through which the wave is travelling doesn't actually move, either, or at
least not much more that it takes for one molecule to bump into the next one
(think of a Mexican wave at a football match, and you'll get the picture).
The vibrations remain vibrations until they come into contact with something
that can hear, i.e. an ear (but *not* a microphone, because a microphone
merely captures some of the vibration and sends it down a wire).
   The faster the vibration, the higher the frequency, the higher the pitch
of the sound; humans can hear from about 20hz to about 20,000hz (although the
more you abuse your ear by pumping high decibel sound into it, the less high
the frequency you can hear).  There isn't much, musically speaking, between
the 12Khz to 20Khz ranges - you would notice the difference if you compared a
song through 12Khz and 20Khz ears, but there wouldn't be much.  It is claimed
by many that we are sensitive, although not actually aware, of sound well
above 20Khz and below 20hz, and this is why professional equipment will have
such a wide frequency response.
   The intensity of the sound wave determines the loudness of the sound (the
harder you strike a drum, the bigger the oscillation of the skin, and hence
the louder the drum - the frequency is unaffected), and sound is
traditionally measured in decibels.  Literally, 0 decibels (0 dB) is
equivalent to an sound pressure level of 20 microPascals, which is the lowest
possible level of sound that your average Joe will be able to hear.  Clearly,
this is a relative figure, as everybody's hearing is slightly different.  The
decibel scale is logarithmic, because that is the way our brain interprets a
change in sound level (for example, the brain reckons that 40,000
microPascals is only twice as loud as 4,000 microPascals; the figure in
decibels represents our perception of it.)
   Now you are likely aware that computers operate entirely digitally (with
the only possible numbers at the lowest level being 1 or 0, one of two
states, on or off).  So how do we translate an analogue vibration into an
internal, digital, package of data?  Well, imagine the sound coming into the
computer on a conveyer belt, and every few thousandths of a second the bit
coming past is chopped off, and measured.  Got it?  That is essentially, the
way a computer samples a sound - a wave file on disk is essentially a large
stream of numbers, each representing the level that was measured in that
particular time interval.  That time interval is what we are referring to
when we talk about sampling at 11.025kHz, 22.05kHz, 44.1kHz, or even 48kHz.
The number refers to the number of times the knife comes down on the wave,
chops off a slice, and measures it; accurate sound reproduction requires a
sampling rate of around 40kHz, CDs are done at 44.1kHz, and DATs at 48kHz.
Generally the sampling frequency is around twice the highest frequency that
can be represented; so if you sample at 22.05kHz, you are restricting the
discernible sound to between around 20Hz to 11.025kHz.  Which is why the
lower your sampling rate is, the lower the quality of your sound.  Of course,
sometimes you actually want it to sound that bit rougher.  Also, if you know
that your sound won't use higher frequencies at all, then it is fine to
sample at a lower rate, and you'll be hard pushed to spot the difference.
   But as you'll know, if you've used Cool Edit or something similar, you
also get the choice between sampling it 8-Bit or 16-Bit.  So what difference
does that make?  Well, if you know anything about binary numbers, you'll
probably be way ahead of me here, but just in case:
   A decimal number is made up of units, tens, hundreds, thousands, tens of
thousands and so on, in effect powers of 10 (10^0, 10^1, 10^2, 10^3, 10^4,
etc.).  So when you write 3252 you are in effect saying 3 thousands, 2
hundreds, 5 tens, and 2 ones or 3 10^3s, 2 10^2s, 5 10^1s, and 2 10^0s (any
number to the power 0 is always 1).  Similarly, a binary number is made up of
ones, twos, fours, eight’s, sixteen’s (or 2^0s, 2^1s, 2^2s, 2^3s, 2^4s, etc -
2, because there are two possible states, 1 and 0).  For example, the binary
number 1101 is in effect 1 2^3, 1 2^2, 0 2^1, and 1 2^0, or 8 + 4 + 0 + 1,
13.
   An 8-Bit number can represent 256 ((2^8) - 1) different states (0000,0000
through 1111,1111), and a 16-Bit number 65,536 ((2^16) -1) different states.
You remember earlier we said that when the computer measures the level of the
incoming wave on the conveyer belt, it stores it as a number.  With an 8-Bit
sampling resolution, it has to choose that number from 256 possible states,
so if the wave happens to fall between 2 of those 256 numbers at that
particular time interval, the computer has to choose the nearest.  You've
probably seen the same thing happen in primitive graphics packages - draw a
diagonal line, and you end up with a stepped line.  16-Bit, therefore,
provides a lot clearer sound quality, as you have more levels to choose from;
even 16-Bit, however, is not perfect, and studios commonly work with 20-Bit
resolutions, which provide 1,048,576 different possible levels, or 24-Bit
resolutions, which provide 16,777,216 different levels.
   Similar to there being a relationship between sampling rate and the
frequency response of the sound, there is also a relationship between the
dynamic range (the possible variation in level of the sound) and the sampling
resolution.  A 16-Bit resolution gives a dynamic range of 96dBs, or 6 times
the resolution.  Don't worry about why, just accept it.  When we say a
dynamic range of 96dBs, we do not of course mean that the loudest possible
level is 96dB, we simply mean the range of possible levels is 96dB wide (any
amplifier can make something louder or quieter quite easily.)
   One thing you should ensure when sampling, then, is that your source is
within the dynamic range of at the resolution you are sampling at.  As an
experiment, shout or scream into the microphone at 8-Bits, and then repeat at
16-Bits.  When you look at the 8-Bit one, you’ll notice that the wave is cut
off at the highest possible point, it is just a straight line or block going
as high as the top of the screen.  What this means is that there were sounds
at higher levels than the resolution allows, but the computer couldn't cope
with them because it was only sampling at 8-Bit; thus it assigned them to the
nearest level, which was the highest possible one.  This is known as
clipping.  Your 16-Bit sample will probably still have some clipping, but
considerably less.  To get round this, either use a compressor, so that all
sounds are restricted to a certain dynamic range, or adjust your gain and
input levels.  If you know you are going to be recording a very loud noise,
drop the gain right down, to keep it all within the range.
   Of course, if you are looking for weird effects and so on, you may wish to
try ignoring the guidelines for good quality sounds; things sampled at low
resolutions, frequencies or with clipping can sound interesting.  It is
important that you understand what they mean, though, as you can only
properly experiment with something that you understand.

   What It Means To Be A Tracker
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
     By Ganja Man/LOK

   Before I start, I'd just like to say that I expect to be flamed for some
of the opinions expressed in this article.  A lot of people probably won't
agree with what I say.  Fair enough.  This is what *I* believe tracking
should be about.  Doubtless, there will be those who have a different
opinion.  I'm perfectly happy to merely ignore them.

   Why Do YOU Want To Be A Tracker?
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
   There are no definitive good reasons for wanting to be a tracker.  There
are, however, I believe, a number of reasons that are not suitable for those
who wish to be trackers.
   Tracking will NOT make you money.  Don't ever think it will.  There *are*
a number of trackers, including myself, who have got recording contracts/game
contracts/whatever through tracking.  The numbers are few and far between,
and if you want to be a music 'professional' quite frankly you'd be better
saving up for some decent MIDI equipment/samplers and making demo tapes to
pass along to record companies.  That route is how most artists get into the
business, and I can't see it changing much.  Tracking is *NOT* about making
money.
   Don't track for respect.  Sure, it's nice when someone e-mails you and
tells you how great (s)he thinks your latest track is, but it isn't a reason
in itself.  Of course, tracking merely to get feedback on your music is
something different; without tracking my music would be infinitely worse,
because I would never have got the insights into what I'm doing wrong.
   Don't expect to become another Necros/Skaven/whoever overnight, or ever.
Very few people become recognised as major trackers, no matter how good they
are.  Most will simply go along unrecognised, doing their thing, good, or
bad, without too many people paying attention.  If you're the sort of person
that is going to be phased by this, then maybe tracking isn't for you.  You
should be happy merely writing the music; if you're not maybe you're in the
wrong game.

   How To Act When You're A Tracker
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
   First of all, above all else, DON'T become a tracker too early.  DON'T
release your first five or six tracks, they will absolutely suck.  By all
means pass your tracks around to friends etc, and get opinions, but uploading
to FTP sites should be avoided until you've got at least half a year of
tracking under your belt.  You may think your tracks sound great; when you
listen to them in two years time, you certainly won't.  I never did.  I made
the mistake of releasing one of my first tracks, and live to regret it to
this day.  Fortunately when I released it the Internet was not a major force,
it just got spread around a few local BBS’s and nothing else.  With things
the way they are now, your tracks could come back to haunt you much more
easily.

   Secondly, take all criticism with good grace.  If someone emails you to
tell you they hate your track, ask them what it was they hated, and you can
put it right the next time.  Conversely, if people write to tell you they
liked your track, email back and thank them.  A number of 'top-name' trackers
merely ignore comments they receive, or at any rate never reply.  Personally,
I try to reply to every comment I get, good, or bad, even if it's just a
short 'thanks for your comment'.  Elitism should have no place in our scene.

   The Ethics Of Sample Ripping
  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
   Sample ripping is a highly contentious issue.  To some, it is a plague
that is out to destroy the scene.  To others, it is the life-blood of the
scene.  Here are *my* views on the matter.

   I do not think there is a tracker in existence who can honestly say
they've never used a ripped sample.  Everyone does it, especially when
they're starting out.  Personally, I see nothing wrong with this.  A sample
belongs to no one person.  When you sampled it, it did not belong to you.  If
it came off a CD, it is 'owned' by the record company that produced the CD.
If it came from a keyboard, it is in the public domain; it neither belongs to
Roland or whoever, or to you.
   There are those, however, who will say that ripping samples is stealing.
I do not blame them for believing this; they have been indoctrinated through
their life into believing that private property is sacred.  They are, of
course, blatantly wrong.  If a sample is ripped from you, are you deprived of
its use?  Of course not.  Was that sample ever your private property in the
first place?  Even by the standards of the capitalist state?  No.  When I
hear one of my samples 'ripped', and used in another tune, I feel proud.
Proud because I have, in some small way, contributed to creating this
entirely new track.  Proud because I have assisted a fellow tracker in his
pursuit of musical excellence.  Those who speak out against sample ripping
claim that they can no longer use their own samples, because since they have
been ripped, they sound too 'samey'.  I would argue that the person ripping
your sample has done you a favour; by ripping your sample, they have stopped
you from using a sample numerous times and falling into a rut where every
track you write sounds the same.
   Some also argue that if everyone ripped from each other, there would be no
new samples.  This is true.  But it is quite clear that everyone will NOT
just rip from each other.  By sampling yourself, you have the chance to use
sounds no-one has ever used before.  Some claim that those who use entirely
ripped samples are just 'sponging' off the rest of us who do sample.  I
disagree entirely; those of us who create our own samples always get first
use of them, and have the chance to create something unique; those who rip do
not.
   In conclusion then, my advice to you would be this: if you hear a sound
you like in a MOD, rip it.  There can be no point in sampling something again
if you will only achieve exactly the same sound.  But when you rip, make sure
you credit the original author of the sample.  It's only common courtesy, and
I personally see it as a mark of respect to those who I rip from.  If you
want your tracks to have a sound that does not exist in any MOD format,
sample yourself.  Simple as that.  The tracking scene is about community, not
any stupid idea of private property.

   Adding Swing/Groove
  ~~~~~~~~~~~~~~~~~~~~~
   By Kevin Krebs

   The traditional method of adding swing to a track is to systematically
alter the speed to produce syncopation:

 000 C-5 01 .. F02
 001 ... .. .. F04
 002 C-5 01 .. F02
 003 ... .. .. F04
 004 C-5 01 .. F02
 005 ... .. .. F04
 006 C-5 01 .. F02
 007 ... .. .. F04
 008 C-5 01 .. F02

   This method works, but forces you to put a swing on every channel.  By
using the Note Delay effect (EDx), it is possible to add syncopation
exclusively to a single channel:

 000 C-5 01 .. .00
 001 ... .. .. .00
 002 C-5 01 .. ED1
 003 ... .. .. .00
 004 C-5 01 .. .00
 005 ... .. .. .00
 006 C-5 01 .. ED1
 007 ... .. .. .00
 008 C-5 01 .. .00

   This also allows for easier "morphing" into and out of the syncopation by
fading between the syncopated and normal channels.
   N.B. ED1 delays a note by 1 tick -- you may need to use greater values
depending on the tempo and speed of the track you're working on and the
amount of swing you want, so experiment.

   There is also another way of adding syncopation to a track that involves
the use of longer patterns and a faster secondary tempo.
   Set the primary tempo to whatever you like.  Then, if you would usually
track in a speed of 06, change it to 04.  Then change the pattern length to
60 Hex.
   Now, instead of treating a single beat bar as 04 rows, use 06 rows.  Every
half beat will come every 3 rows e.g.

 000 C-5 01 .. .00
 001 ... .. .. .00
 002 ... .. .. .00
 003 C-5 01 20 .00
 004 ... .. .. .00
 005 ... .. .. .00
 006 C-5 01 .. .00
 007 ... .. .. .00
 008 ... .. .. .00
 009 C-5 01 20 .00

   This doesn't instantly add swing however.  But by placing notes in between
the beats, it is possible to get very nice syncopation.  This method has one
main advantage - your effects column is free.  By increasing the pattern
length and the speed again, you get the ability to do the same sort of thing
as method two (note delay).
