OctaMED Sound Studio Install Issues & Advice.

[Karlos]

Here's my latest work in progress. I intend to add in some of OctaMED's effects and what not in time as I re-learn those things I've forgotten.

Now how would I go about saving this into a 16-Bit wav on my Amiga?

Right, I just got around to downloading this and having a peek. You appear to be using the "8-channel" mode. What I suspect you wanted to use was the 1-64channel mixing mode, with 8 channels set in the mixing preferences.

Then you don't have to halve the volume of all your samples (which reduces their resolution by 1 bit) and you can use higher mixing rates.

A second suggestion is that rather than setting the volume of the sample in the sample editor, store every sample at the loudest amplitude possible (without clipping) and set a default volume in the instrument properties instead. The reason for doing this is that when the sample is permanently scaled in volume in the sample editor, you lose precision (whenever you make it quieter at least). The mixing routines take the raw sample value and scale it by the current instument/channel/note volume using a greater resolution than your starting 8-bit sample.

[Karlos]

@XDelusion

I've uploaded a 10 channel mix-mode octamed track for you to pull to bits, maybe it will help highlight a few tricks for mix mode: here...

It's basically a quick and dirty recreation of the opening theme to the game "Breathless", which I had just gotten the demo of at the time.

So, it uses 10 channels at a mixing rate of 28149 Hz (on PAL, on NTSC you might find it adjusts it slightly higher). Only 8-bit mono samples are used, so your 030 shouldn't have any problems whatsoever. If memory serves, I wrote it on a lower spec machine than that, or at least something comparable at the time.

The blocks contain 12 tracks but I never bothered using more than 10 channels, they were just handy for temporary places to copy/past sections.

[Karlos]

The tempo is all messed up when I put things into 64Ch Mode, which is easy to fix, though I did want to point it out.

The problem is that you are starting with something in legacy 5-8 channel mode. This was always very restrictive and required the use of SPD, rather than BPM and even the SPD settings worked differently than 4 channel mode if I recall correctly.

Basically, do not ever use the old 5-8 channel mode other than to replay old mods written for it. It has absolutely no benefits and tons of limitations. It isn't even faster than mix mode for 8-channels.

If you are going to create a mod with more than 4 channels, start it out in mix mode immediately. You'll find it far easier to get started. The only thing to remember is that your notes need to be played 2 octaves higher than in 4 channel mode (unless you used Ext samples in 4 channel mode) to achieve the same pitch.

Oh, and why is it that OctaMED SS does not use standard BPM mesures, but instead has that strange tempo interface with the 3 lines? How would I ever sync this up to somthing that is running at say 200BPM, and how would I know the diff between 4 channel mode's timing and that of 64 Mix Mode?

It can use BPM, you get to choose whether or not to use BMP or the old SPD method. The latter is a throwback. If you use 4 channel or 1-64 chan mixing mode, you can use BMP no problem. As with the old 8 channel mode, it's a throwback. Always start new projects with BPM as the tempo control.

[Karlos]

Tempo, in OctaMED works like this.

Assuming you are in BPM mode (which you should be), the main tempo slider sets the number of beats per minute. So in your case, you'd set that to 200.

The next parameter, LPB controls how many editor lines there are to one beat. The default, from the old mod tradition, would normally be 4, meaning you get a new beat every 4 lines. If you wanted to have 1/8th of a beat resolution, you would set this to 8, for instance and space all your beats apart by 8 lines as they scroll up twice as fast as before. If you did set LPB to 8, then your default 64-line block now only represents 4 bars rather than 8.

If you were doing triplet timed music, you would set LPB to some appropriate multiple of 3 (depending on the level of precision you wanted), instead.

The last parameter, TPL is "ticks per line". This value controls the rate of the various player effects but also affects the tempo (the ticks it is referring to are, I guess, the most fundamental timing unit in octamed). The above stated value in BPM is only correct when TPL is at it's default value of 6. The player command 0090x can change this value, which you can put to good effect. For example, if you change it on each quarter beat such that you alternate it between 7 and 5 (or 8 and 4 for a more extreme effect), you get that syncopated "groove" effect that is handy for certain genres. Since the average of 7 and  5 is 6, overall the tempo remains correct.

Hope all this makes sense.

-edit-

Important note: None of the above works in legacy 5-8 channel mode, which has it's own very esoteric tempo controls. I cannot stress it enough, do not use the old 5-8 channel mode!

Here is what I'm talking about.

See in Open Mod Plug Tracker (for Windows), you merely set your BPM, and that's that.

On the Amiga side you got BPM, but you also have TPL and LPB amongst other things.

I understand how to get it into BPM mode, but then I often times need to change TPL, and when I do that, it sometimes makes BPM about useless in that moving the BPM level has no effect accept within the 1-12 range, etc.

All I want to do is set the BPM to 200BPM to match the timing of the other programs I'm using.

[Karlos]

Slight correction to the above. In BPM mode, the default value for LPB is 8, not 4, meaning you do get 8 lines for every beat by default. At some time in the distant past I must have changed my soundstudio.config file to alter that (it's just a text file) to be 4 on my system.

[Karlos]

I should make a few corrections to my post above

-edit-

Done

LPB should be set to 4. I also have this set to that value. Works more like a drumcomputer that way.

Well, in the online help, it does say the default is 8 (at least when using BPM). However, 4 lines per block is pretty traditional.

[Karlos]

@The Goose

The Maestrix does indeed work. I've used it with OctaMED SS on my A1 and also in UAE. It requires that you use mix mode for playback and choose the Maestro as the output device.

[Karlos]

Karlos just pointed me to this old thread. Wonderful resource, thanks.

Cheers

I'll have to get back into it. Not had a lot of time latel

[Karlos]

Since we are reviving this thread, some handy tips (posting from lunchtime at work, so no pics):

When editing/processing 8-bit samples in the sample editor:

1) Convert the sample directly to 16-bit format via the instrument type window.
2) Switch to mix mode, if not already using it, so that you can replay the sample whilst it is 16-bits.
3) Remove any DC bias via the DC offset adjust tool and tell it to calculate the bias for you.
4) If the sample was originally a fairly low sample rate to start off with, use the octave down shift to change the sample rate. Each time you do this, you double the rate. If you want to preserve the aliased sound, do it without any smoothing enabled.

You now have a 16-bit representation of your original sample at a higher rate. Any effects processing you do on it, such as filtering, fading or brightening will be performed at much greater precision. Once you are done, save the 16-bit version off somewhere, in case you need it again.

5) Use the octave up shift to convert the sound back to the original rate (if you wish to do so, you might be happy at the rate it is).

6) Convert back to 8-bit (unless you want to keep it 16-bit and use mix mode).

In my experience this always produces significantly better results than editing sounds in 8-bit mode.

Synthsounds as samples:

The synthsound editor works in mix mode. So, you can create some very complex arrangement of synthsound layering over several channels to produce a fat, rich sound. You might choose to construct a block playing the same sound over a few octaves.

Now, switch to mix mode (if not already) and choose Disk 16-bit. You now have that sound as a 16-bit sample for using without the associated CPU load later.

[Karlos]

Incidentally, did anybody ever try OMSS with the NallePuh utility back in the day? I seem to recall it intercepted Paula audio somehow...

[Karlos]

I'm using The Maestrix with SS on my Amigaone/OS4., It works pretty well. There are a couple of things that will crash the hell out of SS on OS4 that you have to watch out for though.

Probably worth making a note of them. I didn't come across anything so far, but I've hardly scratched the surface of OMSS on OS4 compared to OMSS on OS3.x...

[Karlos]

I'll fire it up this weekend and post back on that. There are a couple of things that cause a total system lock every time. It's a pretty vanilla OS 4 install so I think it's related to SS.

Great, I'll see if I can reproduce them. I should be updating to update 3 sometime over the weekend and we can see if it has any effect.

[Karlos]

I'm always amazed at the lengths people will go to when it comes to continuing to use a music creation system that works for them or that they love. Many guys are still using atari st's with cubase. (I know a few who still swear by it for rock solid midi sequencing)

I'm a hobbyist musician. Investing in something more professional isn't really required.

Having said that, I've used "modern" packages before and whilst they are probably the bees knees for professional musicians, I found them a bit tiresome to use. There's something about the raw simplicity of the tracker that appeals to my developer side I guess.

[Karlos]

Another tip, for production.

You've spent a while creating your 32 channel module and want to render it to disk so that you can convert it for your .

Unless you are using RTG or a high scan rate native resolution, the chances are you've had the mixer running at 27-28kHz ish and without smoothing enabled because even on a fast UAE, that many channels running with smoothing is going to be a pain.

What I always found was that I'd tend to play bright sounds, like high hats and cymbals at notes that had replay rates close to the mixing rate simply because they sounded best at that rate. For example, the "A-5" note for an untransposed sample in mix mode is quite close. This reduced distortion and made the realtime playback nicer to listen to. On a more technical note, it probably causes the sample-rate conversion aliasing frequencies to be closer to the fundamental.

However, you don't want to render your track at 27.x kHz, you want to render it at 44.1kHz or 48kHz, with smoothing for mastering. What you will then discover is that your track sounds kinda dull, like everything has gone through a low pass filter and your shiny high hats have lost their lustre. Soft pads, bass and stuff will sound fine however.

The reason for all this is that at 44kHz, your 27kHz A-5 note is interpolated by the smoothing algorithm. It's also not a rational fraction of the output rate, so the highest frequency changes in the original note just can't be reproduced.

There is a trick to all this. If you can use a mixing rate close to 44kHz when composing, then try to do that. However, if your are using a slower CPU this really might not be an option.

A better trick is to get some high quality 44kHz samples as your "master set" for things like cymbals or other bright sounds. Then create a lower sample rate version (you can use the sample editor to resample them to close to whatever realtime mixing frequency you use) and do your composition. You'll have to set the instrument properties transpose on them so you can keep the original F-6 (about 44kHz) note in your track, but it will replay your resampled sound at around A-5.

Once you are happy, reload the 44kHz version of the sample and remove the instrument properties transpose. When you render your track to disk now, these bright sounds should remain much brighter.