klm09

VST's are programmed using the Steinberg VST API (application programming interface). Basically, it's like a set of rules that define how certain parts of a VST have to be programmed for it to be compatible with the VST standard, and therefore with VST hosts. But the plug ins themselves are programmed in a programming language, such as C++; the VST API itself only defines certain things, it's not the tool in itself with which you make the VSTs. VSTs can be programmed in basically any (advanced) programming language AFAIK. I'm aware of plug ins programmed with C/++ and Delphi, at least. DX plug ins on the other hand have no specific API for the plug ins, you just use the DirectX API, or more specifically the audio portions of it. Again, you program the plug ins in a programming language such as C++ and use the DirectX API as the set of rules to which it has to conform to. It's kind of like diesel and gasoline engines. They both basically do the same thing, but have different design paradigms. Both DX and VST plug ins basically do the same thing, they just use different standards as defined by their respective API's to accomplish that.

jesus christ my guitar would have a shit if I tuned it like that. floating trem is not so good for the alternate tunings. also... are the two consecutive A's in the same octave? that's a little... weird. Yeah, the two high A's are in unison. It doesn't sound weird as such, mainly just.. well, not exactly thick, but richer. I don't seriously use that tuning, but it's fun to play around with sometimes.

Heh, you wrote the tabs, nice. Sites that ripped your work and didn't credit you, fuck them. Anyway, I suppose through deduction of required note ranges on said 4 consecutive strings you could probably narrow it down to a handful of tunings, of which most likely one or two would make most sense. I doubt they'd use some convoluted tuning just so that fingering their riffs would be as hard as possible. But that much is fairly obvious, and I'm not being very helpful, am I? Speaking of tunings, one of my favorites is DADAAd, ie. drop D open power chord. It sounds pretty huge when you play power chords on that on all the strings using just your index finger.

Man, I spent like 15 minutes looking for an answer for you for the hell of it, but couldn't find anything. I wish all bands had like sections on their sites detailing their tunings, gear, techniques.. EDIT: I had a look at some tabs, and some of those say that it's a whole step down, ie. DGCFAd. No idea if that's right, I've never even heard any Neurosis!

Oh wait, NOBODY'S DOING LAST BATTLE ~DECISION~!?! D: D: That's my favorite song from the game as well. I'll do it if nobody else is. EDIT: Wait, ignore my post if I'm making no sense.. I'm kind of tired.

Would it be possible to see a list of tracks currently not being done for this project? So that people could maybe chuck WIPs at zircon?

If you need guitar for the collab, I can help. Or drums.

Also, if you're recording in your closet.. what kind of closet is this, is it full of clothes? Because in any case the acoustics are bound to suck for acoustic instruments.. but if it's full of clothes or other semi-heavy non-reflective material, it might be ok, just really "dead", but probably nothing some reverb can't solve. But if it's hard surfaces, you're going to get a lot of phase cancellation making the recording sound really boxy. Unless you're closet is XBOX HOOGE, but then it'd just be a room I suppose. Anyway, my point is that beyond getting better mics, if you're really looking to improve your sound, working out something to reduce the noise your computer makes might be worthwhile, so that your recordings could come out of the closet.

I just wanted to point out that the idea that MP3 files are inherently of higher aural quality than midi files is false as they're two completely different things. I couldn't be bothered to think and actually answer the questions put forth, I just wanted to point out their factual inaccuracy.

What I usually do when I download a bunch of VSTs at once is I give each a quick spin. I try out some presets and generally muck about with them to see if I can get a good sound out of them with relatively little effort. If the panel is indecipherable or all the presets sound like ass, I usually just delete the VST within a minute or two. There are so many synths, effects, whatever, that even if you "accidentally" delete a few of the good ones, you're still going to end up with a dozen great VSTs for each missed one. Then, when you've narrowed it down to the ones that you really like or think sound good, pick out the ones out of those that do something that the others can't or do it better than the others. Like if you have 2 distortion units, the other covers all the same ground as the other one.. why keep that other one? At least chuck it in some archive folder so it doesn't clutter up your VST menu in your DAW. My opinion is that, if you have truly good VSTs, you don't really need more than maybe 2-3 units of each type, like maybe 2 subtractive style synths, 2 compressors, 2 choruses, 2 reverbs and so on. Hell, if they're really good and versatile, one of each might suffice, but free units like that are rare.

You're talking apples and oranges, there. Think about MP3 this way: Let's say you have a text file made in Word. If you copy and paste that into Notepad, the text is still going to be the same, but you might lose some formatting tags that Word embeds in the text. So if you're using some special font or if you have some fancy tables in there or something, you're going to lose them. At the same time, the file size will probably be smaller, because it's just "plain text" afterwards. The content, for the most part will be the same, but if you copy paste back into word, you will have lost those formatting tags. So the filetype isn't going to make you Shakespeare, or make your spelling and grammar crap either. It's the same with MP3. If you have for example a WAV file, you can encode that into MP3 and get a smaller file in terms of filesize, which sounds basically the same. But what's it's doing is stripping out some frequencies or "information" from the signal, and even if you convert it back to WAV, some information is permanently lost. But the trade off, a smaller filesize for something that sounds almost identical, is often worth it. So MP3 has nothing to do with the quality of your music.

Sound travels in air at about a foot per millisecond, so if you're about a yard from your speakers, that's another 3 ms "latency" right there.

Not quite true, the SNES soundchip did have a built in (rudimentary) reverb effect.

I got my new comp now, and I can currently get what basically equates to 0 ms latency. Running at 96kHz sample rate, with a 64 sample buffer, the latency is under 1 ms. For example Reason just rounds it off to 0. However, to get more overhead, I keep it at 48kHz and 128 samples, which is a bit less than 2 ms. Here's the specs in case anyone cares: M-Audio Audiophile 2496 Asus P5NSLI motherboard Core 2 Duo E6400 2 GB 533MHz DDR 2 (Corsair, I believe) Seagate 320 GB SATA 3Gb/s 7200 RPM with 16MB cache MSI GeForce 7600 GT (fanless / heatsink) Nexus Breeze, 500 watt PSU LG DVD-RW So, basically, the Audiophile 2496 works brilliantly for me despite the PCI-E based nForce 5 chipset motherboard.

Depending on what kind of parts you'd want or need, I could maybe do it. Maybe.

I might be interested in doing a mix of either Dark King from Mystiq Quest, Wily/Sigma from Mega Man or Mother Brain from Super Metroid. However, I'd like to start working on a mix when I get my new comp (which should be tomorrow, but then I should have got it on Friday..). So don't put me down for anything yet, I'll hit you with a WIP to let you know whether I want in, perhaps in the next week or two. Mind you, I have 2 other projects I'm currently part of, and those will take precedence until I'm done with my mixes for those. I'll let you know.

IE Relics of the Chozo? Yeah, I suppose, Prot did a majority of the material for that, didn't he.. So I suppose there's no strict minimum?

Awesome remix. Ja hienoa, lisää suomalaisia! There's more remixes on his site, btw. (URL in the id3 tag).

Two further questions regarding projects. Is there a minimum amount of community involvement required for a project to be instated as an official OCR project? Say if one person did essentially all the remixing / rearranging, and the project, in terms of the community part, mainly just featured other remixers as guest artists, perhaps playing live instruments / solos here and there, or possibly contributing small pieces of the rearrangement, with most of the material (like 70-90%) coming from one person? Second, is there any rule that states that a project has to be released / made public in one go, barring sample or teaser songs? For example, would a project where one song was worked on by the project group at a time (as a band style collab) and then released upon completion, moving on to the next song, be possible?

....

I meant to respond yesterday, but, I kid you not, I loaded up this topic, and then my connection decided to take the rest of the day off. Anyway, I see I've been dropped. Well, I suppose this is as good a time as any to request that I'd like to do Shawn's Got The Shotgun instead, as I was sort of originally intending to. Bye Bye American Pie, well, let's say I can attempt it again later if needed but right now I'm not "feeling" the mix I have for it anymore. I can upload my current mix of Shawn's and link to at the forums, and I believe I can get the rest of it done within 2-3 weeks or less. I have exams next week, but I think I can squeeze in work on the remix in between studying.

The percussion is great, but would be much better yet if a more solid beat came on top at some point, now it kinda feels like it's just one extended build up to the real song. Also, the synths are kind of bland, again, something a bit more on top would make it feel less like background music. Also, consider adding in some sort of harmony or counterpoint. Great potential, but as it is, I don't feel it's up to OCR standards.

Yeah, you're right, it's the latency, not the bandwidth.. I realized the fallacy of my reasoning after I made that post, but.. I thought I'd leave it there none-the-less.

erm, you're kinda missing the point. harddrives are always going to be slower than ram. the point of fast ram is for communicating with the cpu.. so the cpu isn't just sitting there waiting for instructions to pass thru the data bus and stuffs. anyway, i have 667 ram and it works fine. but no big difference. honestly my ram speed is the least of my concerns when i'm trying to make my studio better. cheers. True, RAM is always going to be faster, but I used that example here because for most audio applications, RAM is mostly used for data retrieval as opposed to data storage. Even though the processor might be working hard, it's usually not doing anything RAM intensive (in terms of putting new data in the RAM), but calculating something based on something already loaded in there from the HD, like samples or VSTs, with the result being put in the audio buffer. Of course, the bandwidth is going to matter for said data retrieval, but, if I got my math right, you could stream 5038 tracks/samples at 44.1kHz 24bit resolution before you cap out the bandwidth for a PC2-5300 stick, which I think is a fair bit more than most people are going to ever use. I didn't take into account anything besides the data streaming, but even if only a tenth of the bandwidth was available for that, it'd still be 503 samples at the same time.

Sorry, I don't really have any answers to contribute.. but I'd like to ask something myself as well. I recall reading about there being some issues with PCI cards used in PCIe motherboards, does anyone know anything about this?