Intermediate

Effects -  Panning -  Echoing -  Gating -  How to avoid doubled up channels -  Sampling (CD-Ripping Drives, Filtering) - 

Ok, I know my way around, I can sample and use effects, and I've released some MODs, but just how are certain things done?

Effects

There are a number of effects available in the newer trackers that we didn't discuss in Section 1. Be sure that you are familiar with all the standard effects before you embark this next voyage of discovery. These effects -have- to be used properly, or they can completely destroy what would otherwise be a good track.

Panning

Letís start with Stereo Panning. This is the method by which a sound appears to come from a certain place between two speakers. Panning is accomplished by use of command 8 (In FT2, in others substitute for whatever command they use).
It's a simple command to use. 800 will pan the sound to the far left, and 8FF will pan far right. Values in between these pan the sound accordingly - 880 places the sound directly in the centre, 860 places it a little to the left, 8D0 places it quite some distance to the right.

There is also a Stereo Surround feature in a few trackers. Stereo Surround is actually far simpler than it sounds. Once everything has been mixed, either the left waveform or the right waveform of the stereo pair will be inverted (turned upside down). This effect gives the impression of the sound (yes, you guessed it!) surrounding you.
Stereo Surround works best if you are positioned directly parallel to the centre of where the two speakers are e.g.:

Left Speaker ---------+--------- Right Speaker
                     You

It helps if you are fairly close to the speakers as well. Increasing the distance between the speakers increases the surround sensation.
There is an inherent problem with this method of Stereo Surround though. It only works well if the sound being made surround consists of mainly treble frequencies, since most of the lower frequencies get cut out. This gives the sound a hollow feel. Of course, you can combat this by siting yourself left of the left speaker or right of the right speaker, to reduce the surround effect. But why would you want to?

In most of the newer trackers, panning can also be accomplished through the use of the Instrument Parameters. There will almost certainly be a default panning setting. If you are lucky there will also be a panning envelope.
The default panning has a similar job to the default volume. It sets the instrument to a particular panning position, which gets used every time the instrument is played without a panning command.
Panning envelopes offer greater flexibility over the stereo positioning of an instrument.

The problem with panning is that many people donít know anything about panning theory and how to set up their equipment. Most seem to end up using sounds that swing wildly from left to right. This is agony to listen to! Soft bouncing pans can be effective, but should only be used in moderation.

Virtual Sound Sources Ė By XRQ

As we all know, musicians and music technicians left mono sound a long time ago, simply because the stereo sound sounds much better. The first question is WHY?
In mono there is only one source of sound and, therefore, many problems occur when one tries to put several instruments on only one speaker. It is very difficult to distinguish between them. They practically eat each other and do not come out like they're supposed to.
Stereo brought us two sound sources and it seemed that the problem would be two times easier, however this is not the case. Itís not the fact that there are two speakers, it's just that they can give us many more sources of sound.
The second question is HOW?
The answer lies in (what I call) "virtual sound sources" that are created in stereo sound. Everyone who has ever listened to music notices that some instruments come from far left (e.g. guitars), some from approximately centre (vocals or drums) and some from the right (make up your own example). It is described by saying that the instruments are scattered across the PANORAMA FIELD.
Numerous experiments have shown that a man can tell apart seventeen points in the pan-field. To hear this many he would have to have perfect hearing and years of studio work behind him. We, the common mortals, hear only 11 or 13, if we're lucky. These points are, in fact, my precious "virtual sound sources", because the sound comes from there, and there, and there... But only with two speakers!
The purpose of this writing is to accent the importance of carefully balanced music, of a full pan-field, of a volume of every instrument in that field which we recognise as the music.
So, the third question is - WHAT ONE SHOULD DO WITH THIS KNOWLEDGE? Well, it would be very advisable to look on the pan-settings of your tracker and divide the field onto as many points as you wish (not less then seven). Well you donít have to, I did it for you! That is, if you use Fasttracker 2.0x. Here's the table (hex values): -

7 points

00 2A 54 7F AB D5 FF

9 points

00 1F 3F 5F 7F 9F BF DF FF

11 points

00 19 32 4C 65 7F 99 B2 CC E5 FF

13 points

00 15 2A 40 54 6A 7F 94 AB BE D5 E9 FF

You may have noticed that 00, 7F, and FF are always there; those are extreme points - left, centre and right.
That's it, then. Balance your music right!

Techniques

Echoing

Do you use echoes on various parts of your MODs? If not, why not? They are an easy way of filling out the sound. Really easy to do as well. Simply copy a channel into another empty channel, change the volume of the channel down to under half of its current volume, and insert a row in only that channel. Play back the pattern, if it sounds nice, you've succeeded. Inserting only a single row will only work well at slow BPMs, however, so keep on inserting and playing back until it sounds nice.
One point to remember, and this is something I've seen in many MODs, even ones produced by masters (I won't give any names), is that if the echo is fairly long a few notes will be chopped off the end of the echoed channel when you insert rows. But these notes still exist in the original channel. When the tune is played back the echo will appear to stop at the beginning of each pattern, and then start again. This reduces the 'live' feel of the entire module. Just remember to copy the chopped notes onto the beginning of the next pattern in the playing list, and everything will sound fine.

Gating

Another cool effect (IMHO) is gating. This is usually done with command A. Load a long/looped sample and set it to maximum volume. Now input the channel below (The notes can be anything, but keep the effects the same) (No Volume Column)

   C-5  1 A0F   -   Starts note, slides volume
   ---  1 A0F   -   Sets volume to sample default volume, then slides volume
   ---  1 A0F   -   Sets volume to sample default volume, then slides volume
   ---  1 A0F   -   Sets volume to sample default volume, then slides volume
   ---  1 A0F   -   Sets volume to sample default volume, then slides volume
   ---  1 A0F   -   Sets volume to sample default volume, then slides volume
   ---  1 A0C   -   Sets volume to sample default volume, then slides volume
   ---  1 A08   -   Sets volume to sample default volume, then slides volume
   E-5  1 A0A   -   Starts note, slides volume
   ---  1 A0A   -   Sets volume to sample default volume, then slides volume
   ---  1 A08   -   Sets volume to sample default volume, then slides volume
   ---  1 A06   -   Sets volume to sample default volume, then slides volume
   D-5  1 A08   -   Starts note, slides volume
   ---  1 A08   -   Sets volume to sample default volume, then slides volume
   ---  1 A06   -   Sets volume to sample default volume, then slides volume
   ---  1 A04   -   Sets volume to sample default volume, then slides volume

Now play the pattern, and you should find that you get this choppy sound that gets less choppy with the slower slides. That choppiness is gating. Gating works best when used on strings and vocals, but just play around and see what you come up with.

How to Avoid Doubled Up Channels

Doubled up channels, simply to increase an instruments volume, are extremely bad work. Not only do they decrease the number of free channels, but playback of the instrument will be affected. This is usually due to slight timing errors, and can result in a muffled sound from the mix routine.

There are a couple of ways to avoid having to use doubled up channels. First of all, the not so good ways: -

1) By physically altering the samples volume. This is possibly the worst way of doing it. Altering the samples volume can cause both overdrive from too much amplification, and loss of sample data when individual sample 'snapshots' reach the zero point. Repeatedly altering the volume WILL cause these problems and result in a sample of far lower quality than was started with.

2) By changing the default volume. This may or may not cause any difficulties, it all depends on what tracker is being used. In one like FT2 or PT, the default volume is the same as using a volume command all the time. To explain this, here's an example. You have an instrument that has a default volume of 40 (hex), and you are well into composing the tune when you decide that the instrument would sound better at 20 (hex). You change the default volume to reflect this. But you now have a problem; all the volume commands and slides for this sample were designed for a sample played at 40 (hex). So now the sound gets played back far too loud or disappears occasionally, when it didn't before. It is also far more difficult to get smooth sounding volume slides as you only have half as many volume positions as before.
Something like Impulse Tracker overcomes this problem through the use of a Global Volume instrument parameter. This is a relative volume level, which means that any changes to it do not affect how commands work with it whatsoever.

And now the really good ways: -

3) Using volume envelopes. This is my personal favourite. It works in much the same way to the Global Volume in IT. Therefore IT users and the like can ignore this method completely. This is how this method works. Load a sample, and create a simple two node volume envelope that looks something like this.

Max Volume  - * Node 1
              |
              |
              |
              |
Min Volume  - * Node 2
The first node should be at the top of the graph, and the second node at the bottom. They should be as close together as possible, creating a near vertical slope. Set Sustain on node 1. That's it! Now, to set the samples default volume, simply slide Node 1 up or down.

4) Halve the default volume on loading. Easy, quick, and effective. Whenever you load a sample, change its default volume to half. This can be done using whatever method you like (preferably through Global Volume or method (3)). If you find when mixing that the sound needs more power you can increase its volume, without needing to alter any other setting.
Removing the need to use doubled up channels not only improves the sound quality and mix speed, but it makes it easier to produce a track as well. There's less scrolling around, and you can see more of the pattern on screen at one time.

Sampling

CD Ripping

Do you have a CD-ROM drive? If so, do you use a CD-Ripper? You should do. A CD-Ripper will allow you to get perfect copies of audio on CDs.
You will require a CD-ROM drive and drivers which allow raw data to be read off CDs. Below is a compatibility list that should let you know what drives have this capability.

Drive & Interface

LG/GoldStar GCD-R540C Ė IDE
BTC 36x - IDE

(Ok, so it's a little incomplete at the moment!)

If your drive is listed but you seem unable to get it to read raw data, there may be a few possible solutions.
One problem you will more than likely find in Windows 95 OSR2 and possibly Windows 98 is that CD-Rippers will not seem to work with them. To get around this you'll have to bypass Windows 95's 32-Bit disk drivers by going to Control Panel/System/Performance/File-System/Troubleshooting/Disable All 32-Bit Protect-Mode Disk Drivers. Note that you must have DOS CD-ROM drivers installed for this to work properly.
Certain drives and set-ups will have other problems. One of which is that the first read attempt after every reboot will fail or take forever to start. If this happens, eject and reinsert the CD, and try reading again.

As far as I know, FT2 is the only tracker to have a ripper built in, but it isn't very compatible. If you use DOS for tracking then a CD-ripper called CD2Wav seems to work very well, it'll also take advantage of any 32- Bit CD-ROM drivers installed if you run it under Windows 9x/NT. However it can't rip specific sections of a CD. If you want a small 2 second bite of sound from the end of the track, you have to rip everything before the part you want, which is inconvenient and sometimes impossible.
If you want to rip CD-DA on Windows 3.1, then the only package I know of is Digital Domain. This is quite basic, but it does the job quickly and effectively. On Windows 9x/NT, CD-Worx would be a good choice. CD-Worx comes in separate versions for 9x and NT, because NT uses a different way of handling things. CD-Worx is a nice program, with features for ripping from a variety of CD formats. Audiograbber is the one I currently use, simply because it always seems to work, and you can specify that if any errors do occur simply to carry on. The free version of Audiograbber does have one slight limitation. It can only grab from a randomly selected set of half the tracks on the CD in one session. If you want a specific track, you have to keep on reloading it until that particular track is available!

Filtering

There are a number of features available in most good sampling programs that can be used to improve the quality of the sound. First of all weíll take a look at filters, usually there will be some sort of controllable low/high pass filter that you can use.
At their most basic form, filters are used to remove (filter) various frequencies from the input signal. The frequencies removed may be lower than the cut-off frequency (low pass), higher than the cut-off (high pass), in the range between a low cut-off and a high cut-off (band pass), or outside of a similar range (band stop).
Low pass filters give a sound a deeper, more booming One purpose of using a low pass filter is to remove noise from a low pitched bass sample, it can also add fatness to the sample as well. The most important thing to remember is not to use a low pass which lets too high frequencies through. A low pass of about 8kHz seems to work fine in removing noise from most bass samples.
High pass filters are also useful, and can make very interesting sounds. When used on a bass type sound, they can give it a "hollow" quality. The higher the cut-off of the filter, the more hollow the sound.