Lunahorum

for some reason my pitch bend wheel is having some problems with a variety of synths including Massive and Crystal. In massive, it bends 2 notes down but barely any up. In crystal, it bends more than 2 down and less than 2 up. Any advice?

well I finally tried sonar and reason - havent tried cubase yet. I like them a lot. I am previous a FL studio user. Seems like all three of them pretty much do the same thing. One thing I noticed was reason 4 keeps automation clips grouped with the midi clips they go with which is something FL studio does not. It's also cool how you can put cables in the back of the components in reason. Sonar is sweet too. Seems like they all do the same thing but in a different way. I think I like them all.

yea maybe that is it. I will try plugging the guitar into the mic in sometime.

I have this cheesy microphone I bought from radioshack and it works fine with the realtek microphone input, but it isn't getting any input signal at all when I plug it into the 2496 soundcard. I don't have a preamp (I don't have enough money and I resort to using a noise filter in audacity), but I can play my guitar hooked directly into computer just fine. I am not getting any signal with the microphone hooked in though. I have also tried hooking the microphone into my amp, but still nothing. It only works when I plug it into my realtek motherboard sound card input. The funny thing is that my guitar also works when plugged into the realtek motherboard input. That was badly worded so overall here's the question: Without a preamp, I can play my guitar into the soundcard input and receive a signal no problem. I cannot hum into the microphone into the soundcard input and receive a signal. For some reason, the motherboard sound card "realtek AC97" can have a microphone plugged into it and do just fine. Now this is the standard $5 headset microphone from radioshack, nothing fancy. Just wondering why it gets a signal through the ac97 and not the audiophile 2496 (and yes I switched inputs in control panel - sound and audio settings because I can play the guitar just fine through both) edit even shorter version: if I can play the guitar straight into the soundcard, why can't I play the microphone straight into it? On another note, what exactly does a preamp do? Without one, I am reading around -18dB on the guitar input (The soundcard goes down to -48dB). Would the preamp make it read closer to 0 or do something else? Thanks

When we talk, what makes the words is a changing of overtones caused by the shape of our mouth and stuff. Try singing an "oooo" sound then slowly go to an "aahhh" sound. While you are doing this, you can hear the partials adding on to the ooo. Although the partials are any whole number ratio of the fundamental, I (don't know about everyone else) can only hear the root, 3rd, 5th, and b7th when you go from ooo to ahh. If you are familiar with band pass filters sliding up, it sort of sounds like that. Ok now you know that words are caused by overtone changes, not fundamental note changes. The vocoder has a fourier transform or something in it that breaks apart your voice into its overtones. By detecting the strength of your different overtones and change over time, it applies that constantly changing EQ to the sound you have it hooked up to.

http://h1.ripway.com/max97230/halo.mid Ok there it is I don't know how right this is, but it sounds fairly close.

on it! midi doen shortly

cool thx a bnch!

you need both theory and messing around. For example, it is one thing to know that chorus delays the signal and detunes it creating a "fat" sound, but knowing what that "fat" noise sounds like is just as important.

get a synth and read the pdf that comes with it then try it out. Experiment with the different knobs. Don't be afraid to practice because you might be practicing wrong. Any messing around will make you better, and lead you to the best way to practice.

Drums from hell is sweet. In FL studio, I have "process each output to a different mixer channel" check marked. It only has 8 maximum outputs, but DFH has more than 8 channels (all the instrument mics + overhead + other stuff) Is there any way to send them all to their own mixer tracks? What about in a different program besides FL?

FL: I am using a tube amp vst and I want to fine tune the input. Is there anyway other than clicking on the knob with the mouse to set the value? I don't want to set it with the event editor either because that reads % and not dB. Ok thanks

practice10

just use like a sub tractive synth sound as carrier, but the trick is to change notes midword. I will put up an example in a few days. I have lots of homework right now, but experiment with vibrato and quick scale turns on the carrier.

What is the maximum voltage or current or whatever I can plug into my analog inputs on the sound card. I want to run a line from my amp line out into my computer line in, but I don't want to do it if it isn't in the right range. It says peak analog input signal + 2dBV in my sound card manual. What does that mean? Max Converter Data Width 24 bits. Dynamic range input: 100.4dB (a-weighted) Input Impedance: 10k ohms minimum Ok thanks!!! My amp line out says 8 ohm left and 8 ohm right. Class 2 wiring. Not sure what that means.

what is fl really not that good? What do you guys use then?

nothing beyond piano roll. Music staff is limited IMO though. Just take some notes in a journal while you work on the song like Maj7 here minadd9 here ect so you don't screw up on accident. Or you could compose a song in finale or sibelious or something like that. I don't know though, I really prefer to start straight off the piano roll.

dang. and FL is so good too. They should have like Sonar with FL's midi capabilities.

Access violation at address 03D9B176 in module 'AmpliTube2.vpa'. Read of address FFC00044. that's what I get 2 seconds after loading amplitube into daw. I googled it and it has something to do with fixed sized buffers but I don't know how to change that. Thanks Oh I have FL7 if it makes any difference.

" Don't confuse limiting with normalization... they are really unrelated.. don't even try to make a comparison" I was comparing them in the sense that they alter the amplitude. My previous understanding of a compress/limiter was that only the part going over the threshold got smaller. Now I know the whole wave changes (from sine wave experiment not turning into a square wave at ultra thresholds) thx for understanding

So what is the difference between a limiter clipping the signal and the DAWxc clipping the signal? I tried this out with sine waves to see if I could hear a difference and the limiter is definitely lowering the volume, but not clipping it but only if the release is above 800 or so. I am using mda limiter. Anyways, I don't understand why the limiter doesn't make it sound distorted. It's flattening the wave above the threshold isn't it? I thought that's what it did, but I saved to mp3 and took a look at my super limited sine wave in audacity and guess what, it's just a smaller sine wave that hasn't been flattened at the tops at all. So I guess I still don't understand limiting. o wait I think i just figured out what's happening. A limiter is a normalizer but ONLY at certain spots. It makes the WHOLE wave bigger/smaller when the threshold is broken. For some reason I thought it only crunched the wave at the top where it went over the threshold, but I guess that's distortion or izotope harmonic adder thing. Ok thanks

they have a free demo of it somewhere.

please correct me on everything. Just trying to learn how it works. edit: blue lines in last drawing are not 0db, they are threshold of limiter. maximum amplitude line not drawn.

it's possible to be distorting during the 15ms attack time before it gets down right?

so the attack time is how fast the limiter limits the amplitude, and the release time is long it limits the amplitude after the initial limit?