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.
(Taken from CU Amiga May 1994 - Slightly edited to be more generic)
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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).