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Posts by Ahrion
Ahrion's Profile - Posts by Ahrion
Pages: « 1 2 »
Since yesterday, I'm using C700 in FL Studio. It works "fine" but I have one major problem:

When I record a .spc via "Set Recorder..." and play this .spc file with the SPC700 Player, it occurs that the file stops at a certain position and holds the last played notes until the end (when I skip that moment, it's just silent after that point). At first it worked fine but eventually I changed something in my project (I changed nothing in the recorder settings).
Record Start Pos and Loop Start Pos are at 0 and Record End Pos is at the End. Even if I change that value, nothing happens.
I have a few basic questions:

I have composed my own piece of music with FL Studio and the C700 VST plugin. I converted it with the SPCtoMML converter and loaded this txt file together with the samples into a rom with AddMusicK. That worked out quite well.
But now I want to change something in the txt file (Pitch slides were not taken over). My problem is that the C700 plugin does not export to a spc in an optimal way: The samples in each channel are changed continuously.
It makes no audible difference, but the txt file is completely chaotic and therefore the txt is almost not editable.
Is there a better way to create a MML file for AddMusicK, for example to convert a MIDI file into MML where I only have to insert my samples?

Another problem I have with this track: When I put it in a level, all game sounds (jumping, Super Mushroom etc) in this level will be played with a higher pitch.
Thanks for your answer!
I have just converted a Midi to MML with PetiteMM, added samples and instruments to it and added the respective instruments in the individual channels with @. The txt file is now nice and clean, but now I have 2 (or 3) other problems:

1. all my instruments are extremely pitched. I have already read that I have to change the value behind the $ for this. How do I know which value to set (and behind which $ sign)? I set the last value of the instruments to 00 just for testing, then the pitch changes but it's still wrong.

This is how the Instruments look like:
Code
#instruments
{
	"Hit.brr"	$FF $E0 $00 $01 $81	; @30
	"Kick.brr"	$FF $E0 $00 $03 $4F	; @31
	"Slap_Bass.brr"	$FF $E0 $00 $04 $A5	; @32
	"HiHat.brr"	$FF $E0 $00 $02 $8D	; @33
	"Bass.brr"	$FF $E0 $00 $01 $8D	; @34
	"String.brr"	$FF $E0 $00 $03 $93	; @35
	"Snare.brr"	$FF $E0 $00 $01 $43	; @36
	"Crash.brr"	$8B $2E $00 $02 $51	; @37
	"OHiHat.brr"	$8A $3F $00 $01 $BC	; @38
}


2. I got an error because i wanted to skip 5 octaves at once and that was probably too much /too high (from c2 to c7). I suppose that this bug will automatically fix itself when the first problem is solved? Strangely enough PetiteMM converts increasing an octave with < and decreasing it with >, although in the AddMusicK ReadMe it says the exact opposite.

3. the song breaks off about 3 1/2 beats before the end and starts looping again from the beginning.

To your previous questions:
1. yes, they are custom samples, so they are in a separate folder. (6 of the 9 samples are still from SMW)
2. in channel #6 there is no command, in channel #7 there is a (!1,-1), (!3,3) and (!0,0). At the beginning of the txt file is
Code
(!1)[$F4 $09]
(!2) [$ED $80 $00]
(!3) [$ED $47 $1C]

3. I do not think so.
4. I use bsnes 1.15
Thanks a lot you two! You were able to help me with all my problems. Actually 3 of the channels were too short. The instruments are also correctly tuned now.
I actually misinterpreted the thing with the pitched game sounds. The sounds were simply overwritten with my samples because I forgot a #default in #samples{}.

In any case thank you very much! This community here is great!
I'm pretty new to AddMusicK. In my Midi file I had a portamento effect (constant 20%) between 2 notes (from o6 f+24 to o4 d+32). I tried to reconstruct this with $DD $XX $YY $ZZ. Unfortunately it doesn't sound the way I want it to sound and I can't find an explanation for $DD. Can someone help me here and tell me how to use it and what the different hex values mean? Something like start note, end note, speed etc.?

This is how it looks at the moment:
Code
f+24 $DD $00 $20 $90 << d+32 r96
I have a question, but I'm afraid you can't really fix it. Is there an upper cut-off frequency? I know AddMusicK gives out an error if you go above o6 A when the pitch is too high. So far I've been able to work around this by setting the octave lower and simply tuning the corresponding sample an octave higher.
But now I have a sample that overflows at a certain A and then changes to the lowest octave.

Can I change anything about the sample itself? Or is this simply a hardware limitation of the SPC700?
Thanks a lot! I will try that out.
Unfortunately I get an error in brrtools (size of file isn't a multiple of 9 bytes)
Also I want to play the sample one octave higher. When I downsample it I go in the completely wrong direction.
To test this I halved the sample rate of the sample in FL Studio with the C700 plugin and then moved everything up an octave. Acoustically I am then exactly on the same level as before and the sample overflows again.
Originally posted by Unsaved Map16
BRRTools doesn't take into account the 2 bytes header that Addmusic samples have (both when encoding and decoding), so it's better if you use another tool. For example, use brr2wav included in Split700.


I have already tried this, but the wav that results is silent.

Originally posted by Unsaved Map16
Try downsampling the wave with a different program and then import it in C700. For example, I usually use OpenMPT for that (with which you can also use C700), I don't know much about using C700 in FL tbh.


That was my original plan. Since I can't get the sample converted into a wav I tested it in C700.

Btw this is the sample: https://www.dropbox.com/s/3x1hm3polpmhc89/ed2-113_08.brr?dl=1

Maybe you are able to convert it to a wav file? I would do the rest by myself.
That's the same output I got. God I'm stupid, should have tested it in OpenMPT and not in windows media player.

Now everything works and the sample sounds correct!
I have changed the global song "Stage Clear", which now contains an additional custom sample. The SPC player plays the song correctly, but in the game the custom sample is either played incorrectly or not at all.
In a level with an original SMW song the sample is not played at all, in a level with a custom song the custom sample from the local song is played instead of the custom sample from the "Stage Clear" song.

Does the sample have to be placed somewhere else than usual? And is the samplegroup crucially for this?
Thanks a lot. :)
Is there a list, which commands exactly save data size?
e.g. loops save data size and comments or the lenght command l does not change the data size at all.
My main goal is to optimize my MML code without changing the audio.
Thanks, but are there other commands beside (super)loops to save data size? For example the q command?
So remote codes also save ARAM? Didn't know that, thanks!
But other than that, (super)loops seem to be the only commands to reduce ARAM (without changing the Aduio)? Good to know.
As the title says, I'm looking for a specific sample. It's a typical 80's Snare sample, like the one in the Intro of Rick Astley's Never gonna give you up:

https://www.youtube.com/watch?v=dQw4w9WgXcQ

(No, I'm not Rick Rolling you here #tb{^V^} )

It would be very great if someone can help me find it (.brr or .wav) or to recreate it somehow.
I want to change the ADSR values of a sample (mainly the A value) in the middle of a song.
Before I get to that, though, I have a general question about the $ED command.
In the ReadMe the following example is given:
Code
For the ADSR command:                   $ED $24 $A9
For the instrument syntax: "SampleName.brr" $A4 $A9 $00 $XX $YY

My first question is where does the difference of 8 in the D value come from? (Does it have anything to do with Gain?)
Also the table goes only to 8 instead of F so why is the D value in the instrument syntax always higher? Is the difference always 8?


Now about my actual problem.
I have the following sample:
Code
"Strings.brr" 	$FF $E0 $00 $06 $30


There are a lot of short notes at the beginning of the song, so I want to keep the high A value. Later, there are longer notes which I would like to play with a slightly lower A value (for example B).
My approach then was:
Code
$ED $8B $E0


The sample then didn't play at all unless I set the S value to F but even then the A value was really low.
I then compared the sample with the following $ED command and instrument syntax and they sound different:
Code
"Strings.brr" 	$FB $F0 $00 $06 $30
$ED $8B $F0


I have the feeling that with the $ED command you don't set absolute values at all but add/subtract them to existing ADSR values, is that right?
How embarrassing... works fine now, thanks!
I've been wondering for a while how exactly the free ARAM found in the stats folder is calculated.
A quasi empty txt file without samplegroup and with samplegroup #default gives 96E9 (38633 bytes). With the smaller samplegroup #optimized I get C980 (51584 bytes).

First of all, I wonder why there is no difference between no samplegroup and #default, since if I don't set a samplegroup I should have more space (of course the ingame sounds are buggy then).
The values I get when I set the samplegroups make sense when you look at the difference in their sizes.

What I don't understand though is that when I work with e.g. the samplegroup #optimized, AddMusicK shows me the Echo ARAM error at about 12kB Free ARAM. Is it coincidence that about the size of the #optimized samplegroup is missing at the end?
Thank you for the information!

By global song data, do you mean the global songs themselves? I think I removed a global song once for testing and nothing has changed on the free ARAM.
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Ahrion's Profile - Posts by Ahrion

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