SDK: UMX Question

[~][FGS][Nobody~]

COS MUSIC PLAYING IS A CLIENTSIDE THING!!!

[ShadowRunner]

I think I know what is "clientside". I think opposite is "serverside". To switch on my computer I press the big button?

[HDD21]

I find the file size is too large and reducing sample rate degrades the music, especially the music I wished to use. I ended up with a 14MB file for a 10-minute track with reasonable quality, mind. But 14MB is too big, It is just experimentation.

Nobody, you have said that module files contain representations of the instruments used, which is how i also thought it worked, wouldn't there be a way to output tiny 'blocks' of the audio to a corresponding instrument in the correct digital format, thus converting the source file to a 'real' module file. It sounds like it would be a big job though because of the sheer amount of audio 'combinations'.

Also MP3 is a 'lossy' format, that when converted to WAV must end up replacing the 'lost' bits with something non existent,? maybe some way to strip those bits from a WAV File? also I do know they are both encoded differently, but is MP3 also compressed in the sense that it has to decompress it before playing, rather than just stripping unwanted 'inaudible' bits from the audio? Not counting the different encoding that it uses, as all formats encode it differently, but does MP3 actually compress/decompress the audio on the fly as part of this? Also I know the bitrate of WAV is very high, say 1000kbps or uncompressed, so why not reduce the WAV bitrate to say, 320Kbps before importing? I did try this in Mobile Media Converter but that change refused to register, so I wouldn't mind experimenting this technique using a couple more programs. I do not know if I am missing something here, I know a different format, is a different format, but trying to find a compromise between them especially because if an MP3 is say 256Kbps why not reduce the WAV file to say 320-500kbps, the little extra to compromise for any unwanted results that may occur, since a WAV file must use more data per sample of audio than an MP3 and similar files do... Or is PCM just stuck at high bitrates/uncompressed mode by it's sheer design? For those who do not know, PCM is the encoding used in WAV files, compact disc audio, etc.

Everyone talks about changing the sample rate, but that degrades frequency response, too. Anything under 12,000Hz seems to degrade frequency response but also induce artifacts... people whom i know seem to think sample rate is frequency response but I thought sample rate was how many times a second a 'snapshot' of the audio is taken, I do know it would have a direct bearing on the maximum frequency to be recorded, of course but to reduce sample rate to any good size for server downloads reduces it too far when maybe the bitrate could be reduced instead...

[~DJ~]

COS MUSIC PLAYING IS A CLIENTSIDE THING!!!

JUST MAKE A DLL AND DO HAX LIKE ALEX DID OR SOMETHING?

[ShadowRunner]

Appreciate all this guys.

1. Quality is not that much of an issue, it's never going to be as good as DXHR, however hard we try. We've cut corners elsewhere.

2. I've managed to get tracks about 1:30 or 2 minutes long to fit into umx and loop, but the main theme, Icarus-Mr Neyrus remix, is about 4 minutes long. Still could do with getting this important track into a umx.

I find that sometimes one of the two Modplug steps is irrelevant, a umx can be made straight from wavepad, once it's .it file.

Ultimately I don't know how to use Modplug Tracker well enough to extend the track length to fit the music in.

[Magus]

2. I've managed to get tracks about 1:30 or 2 minutes long to fit into umx and loop, but the main theme, Icarus-Mr Neyrus remix, is about 4 minutes long. Still could do with getting this important track into a umx.

I recall you asked for the author's permission to use that one... perhaps you can contact him again and see if it's available in a tracker format?

[ShadowRunner]

I can duz that yes, he's actually contactable, thanks.