OctaMED Sound Studio Install Issues & Advice.

[XDelusion]

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]

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.

[x303]

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.

LPB should be set to 4. I also have this set to that value. Works more like a drumcomputer that way.
And don't forget the maestrix http://aminet.net/util/libs/TheMaestrix.lha. This program allows you to use AHI with SS.

[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.

[Vegas]

Hi! Thanks a lot for the information! This threat solved a lot of Octamed mysteries for me.

[TheGoose]

LPB should be set to 4. I also have this set to that value. Works more like a drumcomputer that way.
And don't forget the maestrix http://aminet.net/util/libs/TheMaestrix.lha. This program allows you to use AHI with SS.

What!? Does this really work?

Based on Richard Korber's work, The Maestrix emulates the functionality of his
Maestix MaestroPro API library, but instead works through AHI. Now any tools
you have which can play using the maestix.library can now be directed through
AHI. As an example of the possibilites, SoundStudio can use this library, so
now you can use SoundStudio to play multi-track 16-bit modules through AHI!
Yes, now all you AmigaOS4 users can play with SoundStudio once again, and
catch up from where you left off without the need to wait for an OS4 native
version. It also works with standard 4 tracker modules as well, so you don't
have to worry about not having CIA timers, Paula hardware, or how SoundStudio
doesn't directly support AHI. Now it can!

[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.

[Wilse]

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

[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

[magnetic]

Thx for reviving this thread. Great info. I never head of Maestrix. This should have been a news item!

[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...

[spihunter]

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.

[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...

[amigadave]

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.

Great tips Karlos!

Just thought I would quote them and give thanks to bump the tips back to the top of the thread and front page.  This kind of stuff is very helpful to people like me that have little to no experience with editing samples, or original works.