hi guys i wanna share some really cool tricks which seem to be really obscure no idea why lol

ok first one if you're using a wavetable chip like the n163, pc-engine, etc or a sample chip with very little channelcount like the amiga, you can make a non-fixed chord setup by having a waveform that's a perfect fifth and a waveform that's the same sound as a singlet. these can be pretty short, a perfect fifth is 3 periods of the higher note or 2 periods of the lower in length, and most chords a lot of people use have at least one perfect fifth they can reduce with, if not two (eg for "standard" seventh chords). you can therefore make up to a 4-note chord only use 2 channels without having a prerecorded chord sample or using an arp, or do a really big chord with one of these and a 3-point arp in one channel.

ok thing 2, having a very slight vibrato running on at least one channel that's harmonizing with another can prevent them from phasing each other out, without needing to be picky about notes. this isn't usually an issue with any live instrument since they all (even synths) do it kind of naturally, but in chipping tunism it can b important

ok thing 3, if your system of choice has a noise channel and you're not using it for like a percussion sound, you can use it at very low volume to slightly meat up sounds, play around with a min-volume noise moving around with a 3-part pulse-pulse-tri piece. mr hezed has done it a lot and i have used it sparingly, but since i also use noise channels for percussion p heavily it's hard to notice im doing it since it becomes a tail to the percussion

ok thing 4, specifically for tracker users: if u use a crapload of variant instruments and properly use your channel effects u will usually do a better job making hyper-expressive music than u will turning the speed to 1 please stop forcing yourself to use unreadable modules, figure out how much space u like to have in ur module for readability and think of making your modules faster if u find urself bashing ur head on the tracker, dont go straight 4 the vinyldevil cosplay i beg of u

these are great tips, especially the wavetable one!

the only one that i truly get is the third tip, i will make use of the other tips when i hopefully reach the day where i understand these clearly!

Yeah, any chance we can get a video demonstration of these?

If that isn't too demanding.

don't forget echo (games from konami and namco used it a lot), surprise notes, chords from notes from different octaves (games from konami used it a lot as well), etc. - most of bgm songs from these games only used 2 channels, leaving the 3rd channel empty for sfx, so they were plenty of dyad chords

general tip for dealing with different hardwares: dont try to make the system work with your song, try to make your song work with the system.

if you wanna aim for a perfect "what i hear in my head is what i want as a result"-experience, just use a DAW and make music in it.
to me, an important part of writing for different systems is always to adjust myself and my music to that system.

of course, the better u get to know a system, the easier and more natural it gets to have very clear ideas in ur head on what u wanna achieve, so the "i hear it in my head already"-approach might work for that too at some point.
but generally, sometimes its better not to try to run through a brickwall with your head first only to get that one specific detail the way you want it, but rather look for what the chip or hardware is able to achieve and follow its lead.

An example module showing the first 3

number 4 is hard to demonstrate with a module doing it, so instead go try to read a module not doing it (youtube dot com user vinylcheese i have no idea how he can read those damb modules its impressive)

My patented strategy for good-sounding echo:

* Reduce volume (obviously)
* (If possible) stereo pan the echo.
* Delay the echo by an amount that avoids the echo and the original channel beginning a note at the same time a lot. (Unless emergent dyads are an important part of the composition.)

(I treat echo volume, echo panning, and echo delay as being on a triangle - you can't increase one without decreasing the others. Otherwise you'll end up drawing more attention to the echo than you do to the original channel.)

* Detune it down a little bit (if $80 means no detune, I go for $70-$77)

* Put vibrato on it. If you also have vibrato on the original, make the echo's vibrato deeper and slower. (It doesn't matter if it sounds weird on its own - the listener will never hear it that way.)

(It can also help to have the two echo speeds not have any common factors (like 3 and 5) - this means that the vibratos won't "sync" as much, so the two channels should null less.)

* Optional: Edit the echo instrument so it doesn't have as many high frequencies. Ex: for NES, if my lead is a 25% pulse, my echo is a 50% pulse. With FM you can get really subtle.

And you're done!

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With a chip where your three stereo choices are dead left, dead right, or dead center: You can get creative with the ways you try to balance the panning.

Say you need more sound in your right ear, bit not a full channel's worth. Well, if you have, say, a consistent eighth note pattern in one channel, pan 70% of the notes to the right and 30% to the left.

My ears find that a channel jumping from left to right is a lot less distracting than jumping from center to right. And since this channel now spends most (but not all) of its time panned to the right, it can serve the same mixing effect as soft panning the whole channel to the right.

And you can experiment with how you scatter those left-pan notes among the right-pan notes to add "rhythmic" detail without actually adding rhythm. In fact, sometimes I do this with a 50/50 split between left and right, just because it sounds cool.

Also, feel free to fiddle with the panning of existing elements as others come in, to keep both ears relatively "full" of music content.

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I posted the module for my Summer Chip entry in the BotB Discord

a while ago, which makes use of both of these techniques.

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_{(Additional "soft hard-panning" technique: You can start a note panned to the}
^{center, and then change the panning to left/right mid-note.)}

I concur regarding speed == 1. You want the greatest FPS (frames per slot, a.k.a. “speed”) that covers your finest granularity of note length—excepting flams, trills, or other anomalies better deferred to effects (SDx delay, Jxy arpeggio, etc), while keeping the tempo near its midpoint (0x90 with IT's 0x20~0xFF).

3 and 9 are good starting points, as they are divisible by both 3 and 4 at 4-slot increments. (In other words, you retain the ability to form perfect triplets.)

If you must make a fast song in classic tracker mode (= S3XMODIT), unless you're going for PC speaker portas or want to emulate WB3 coin drops you're better off with average ticks and high BPM than low ticks and average BPM.

Just to demonstrate. Shown here is the same slide at A02/T40 and A06/TC0

The difference is audible too.

In addition to slide accuracy, you get ultra-high-precision SDx, SCx etc. commands for your experimental needs. The downside is having to do arps manually as Jxy (or 0xy) becomes useless at high speeds, but if you're making fast music you've probably gotten used to that anyway.

Ironically, OP is referring to a FamiTracker/Defle composer while the tick_speed=1 thing is even more relevant to S3XMODIT formats since they skip the first tick, which breaks, like, every effect.

i mean i was just talkin abt readability but that's a super cool thing to know tbh

im gonna write some nice aym chip for the furnace users out there that wants to get into aym or already do them on furnace but struggle to:

General Tips:
- since this is a 3-channel chip, avoid wasting polyphony unless necessary or you need to.
- the envelope is very good. abuse it. the writing about it is down there.
- the noise could be hard to use but there is a really creative way to expand polyphony
- try not to use macros a lot except for the basic stuffs (volume and arpeggio)
- since it's an aym i will encourage you to pick the ym2149 with 1.79mhz clock. (good range of noise and better envelope) though i won't stop you if you want to use the 1.77mhz ay-3-8910 for the zxs128k type stuffs
--------------------------------------------------------------------------------
ok heres the noise stuffs first

AY NOISE TIPS
to expand the polyphony by a bit i suggest you layer your arpeggios with noise (by turning on noise and square in the same channel, for volume control on the arp+noise hybrid i'd say you should use hihat and crash volume control on it. it doesn't sound that bad.

mind you that noise control on teh ay is all global so manage your noise freq wisely or else clashing happens. (you don't want that to happen, do you?)

heres the envelope part (this might be long)

AY ENVELOPE TIDBITS
ok do you remember the multiplier settings in the fm side of stuffs if you do fm before? good, because that's what autoenv num in furnace is. you basically have 16 levels of envelope auto-pitching control and its sort of related to the current note you're playing in some ways. for example, setting it to 1 will sync it up to your current note the best it can and setting it to 3 will be 18 semitones above your base note. if you're lost on what the multiplier ratio is, you can open a separate window with an fm system and editor on and check the pitch by solo the operator. harmonics are really helpful because it helps create a stronger sound design choices for the bass on your ay track. another way to use the envelope pitching is manually by a command (23xx) in furnace. the higher the xx value is, the higher the pitch of the envelope is. the advantage of that is you can go completely wild with it and have some really cool effects to accompany the song, albeit be aliasing from the envelope and square or a really rough and hard bass that could as well fit in a dubstep track.

as a sidenote, autoenv den is useless. it basically throws all of the tonality of the envelope off and make it a mere simple hardware volume control. (not recommended at all, use volume macro instead.)

now if you want to take use of the autoenv num to its fullest extent, you have to set the envelope to play with the square (do it the same way you set the noise and square to play at the same time, except change the noise with the envelope.)

and lastly, there's envelope settings. this is used to shape the ay envelope to 3 different shapes. enabling it will make it play in a |\|\|\|\ manner, which is the default. turning on direction will shape it into a triangle instead of a saw: /\/\/\/\, and lastly setting it to alternate will shift directions of the envelope (only affects saw as mirroring triangle resulted in nothing.): /|/|/|/|

that's all the writeup of this aym tips for the furnace users.

REMEMBER: AY is limited. be creative with all the things you do and of course don't forget to have fun doing it. It's like a mini challenge after all. :D

some s3xmodit tips

echo tip: reduce the volume of the first echo layer to 25% of the original volume, and then halve the volume on every next layer. so, for instance, if you have an instrument at volume 64, the next layer would be 16, then 8, then 4, etc. this makes the main layer more distinct.

another useful tip: lowering the volume of instruments to about 12.5% of the original volume instead of stopping the instruments makes your track sound fuller

That's a good point, Lia. I really like that visualization, though. Did an ffmpeg filter make it?

A drawback to extreme BPM is running into a wall when you want to modify tempo, interactively or otherwise (if you want Mario to skedaddle when there are less than 100 Mario seconds on the clock, for example).

> Did an ffmpeg filter make it?
Not even that fancy, just the foobar2000 spectrogram. Easy to use, general purpose and does the job 99% of the time, for me at least.

it's never a better time to learn to track at speed 16

Favorite of mine for XG/GS drumkits: layer the ride cymbal and ride cup to create a hella realistic ride cymbal sound. You can then slowly switch between which is louder than the other to sell it as the real deal. I'd recommend listening to jazz to get a feel for how it's supposed to sound, you can usually clearly hear the ride there. I have other XG specific secret sauce recipes in development so stay tuned.

NSFsecrets

THE RECIPE TO A GOOD NON-SAMPLE DRUM
- use an audio reference if necessary
- start with volume if you're working with a noise drum. get the volume macro to the point where it follows the trajectory of adsr as accurately as possible. adjust these along the process. triangle only has one (two, based on dpcm usage) volume level(s).
- use the arpeggio and cycle macros as needed. some drums, such as the snare, kick, etc, have a crisp snap to them that just using one of the noise modes just won't be enough. for example:
---> for a snare, have it so that it snaps from the crackling noise mode to the white noise mode within a matter of milliseconds so that it emulates that snap.
---> for a kick, have it drop sharply in pitch once played. if you're using a noise instrument, use arppegio macros because there are less frequencies, more spaced out from what the pitched channels use. likewise, if you're using a triangle, use pitch macros due to this difference.
- the special spice: to make it sound just a bit more better sounding, add a slight vibrato to it after every note trigger. works for a lot of things, surprisingly!
- tinker with your drum often! if your drum doesn't sound right, go in there and play with the values a little bit.

OTHER TIPS FROM THE TOME
- use arp macros as much as possible! while 0xx is very convenient, it is locked into triads due to the 2 possible values to enter into it. arpeggio macros allow for non-looping arpeggios, 7ths, 9ths, 11ths, etc, AND can go below the base note, so utilize them to the best of your ability depending on the track you're trying to make.
- experiment with arpeggios often. arps are not just for making chords, it can also be an effective method of making instruments more interesting. try using an arp on your lead next time to get a little more oomph out of it.
- when using expansions, consider whether or not it fits your needs specifically. many expansions, while amazing and tempting at first, have their ups and downs, differing in specifications and hardware among each and every one of them. think about what will work and what won't.
- have fun :)

and arp macros are great for tremolos, slides, and natural harmonics! :) (btw, a question about music theory: is there any difference between slide and portamento?)

Tip for Yamaha FM Soundchips

TL changes are pretty cool but you can also use FM Multiplier changes for cool effects! It opens up a whole new world of FM sound design!

In action

In OPN soundchips, when used with SSG-EG it can serve as a high-pass filter kind of effect
example

You can also advantage of Furance and Bambootracker letting you have FM parameter macros and make use of quick multiplier changes which can go BONKERS
example

The examples are on OPM and OPN but it also works on the other yamaha chips like OPL and OPZ

I know this thread is dead, but I wanted to revive it to talk about s3xmodit (particularly *.it), openmpt, and the XXk module formats (again, particularly *.it), even if it's not traditionally considered chiptune.

I recommend you use this trick on 16-bit and 8-bit samples which don't sound as noisy at low volumes (typically characteristic of sounds with a lot of strong harmonics and overtones. The closer to a sine wave it is the worse 8-bit samples sound.)

Boost the sample volume of your module to -0.03dB and set the sample volumes to taste.

Now, amplify your samples by Sample Volume / 64 as a percentage and boost them back to 64. You can do this incrementally by multiplying the the sample volume by 2 and amplifying the sample by 50%.

And... Congrats! You've saved space on your song. My game's WIP soundtrack went from around 14MB of 129 samples to 12.9MB of the same amount. Considering a lot of these are pretty long samples, I'm not super surprised, but still.

So, how does this work? It all has to do with the IT's compression algorithm. The way I understand it, by amplifying the sample, you're limiting the actual resolution required to reconstruct the sample, which makes it easier to compress. This is most noticeable by amplifying any 8-bit sample by a ridiculously low percentage and normalizing it. You lose a fair amount of quality. If done correctly, however, it's incredibly hard to spot.

Trick no. 1 in the original post is something I used extensively in my Battle of the Karts entry - the trance stabs and pad samples were harmonized in fifths, the FM piano patches were harmonized in fifths and fourths (for added freedom of voicing).

There's an analogous method for making clean major/minor 7th chords in Metal - have two guitars playing two power chords a major/minor third apart from each other (i.e. E5 and G5)

idk if this is a good tip but: when fds is allowed (nsf+), i usually just make a fifth wave. it saves 2 channels sometimes

oh yeah I've heard gameboy music that does that methinks

I can think of a tip! (Idk if it's good tho) Try to use bass to play as a both melody, and as harmonisation to the rest of the arrangement, it kinda sounds weird when you isolate it from the rest of the instruments, but it can create some nice harmonies when played with for example the chords and the lead. And also try to use smaller number of channels as having to write 20 different tracks for 20 different channels can get messy (at least, for me it was difficult to get my head around bigger projects, you may have a different workflow).

So... 02FD is just gonna casually mention working on a game soundtrack in Impulse Tracker format, and we're gonna let it slide?

openmpt tips:

when downsampling, do not use the r8brain filter, use no interpolation (None), usually keeps the crisp of the sample while still saving some space.

also, try avoiding openmpt for s3m, as it does not support channel panning options etc that scream tracker 3 has. openmpt saves every s3m module in a way where everything before 8 channels will be panned to the left, and after 8 channels will be panned to the right. scream tracker 3 has special options for that.
if you decided that you will use openmpt for s3m, do not use large samples, they will get screwed up in scream tracker 3 and replaced with random disk data (you might get your passwords and personal data leaked xD).

for sample loops, if you're doing forward loops (on), try finding a pattern of waveforms (or a waveform) that repeats and set that to loop. if you're doing bidi (ping-pong) loops, try finding a place where a waveform replicates itself perfectly on both sides (where the side that goes up is the same or similar like the side that goes down) and set the loop marker on the place where you see it replicating.
so, like, if you have a triangle wave that's going like this:
/\
then you set the loop marker at this point:
/|\
because it's gonna play back the other half of the triangle wave when it goes to reverse. you can use this as a trick in reducing sample sizes in your 4k modules etc.

instruments. if you feel like your instruments fade out too early, you can set a point a bit above the down limit, make another point and put it to 0 volume, then just drag it out a lot.

distortion: other than just nuking your sample by amplifying it to 32767%, you can distort your samples by converting them to unsigned and amplifying them by 90-70%, depending on how much distortion you want on your sample, and then convert it back to signed. you can also overamplify it in order for the sample to lose data, but I haven't used that.

highpassing: you can highpass your samples by making a duplicate of them, downsampling them and inverting them, then just play both of these at the same time. you can decrease the highpassing by reducing the volume of the inverted sample... or just downsample it more.