Distortion on a tube amp is overloading the tube, thus bringing up the floor to a higher level while keeping the peaks at the same level, resulting in a natural compression. Solid state distortion circuts try to achieve this effect by essentially re-amplifing the already amplified signal in the preamplification stage. This is why tube amps produce odd order harmonics with distortion and solid state amps produce even order harmonics with distortion.
There's the entire quote that you misquoted. It states that no matter what guitar distortion you are getting (tube or solid-state) it is naturally being compressed. That's basic audiophysics.
By the way, your third post starts the argument. It shows your ignorance and arrogance and how you ask a question and then already seem to have to answer. It first shows up in your second post, but is very apparent in your third.
By the way, you've been recycling for a while now, and now your just taking yourself to a whole new low.
Yamaha NS-10's are a standard because of how accurate they are. And then you're making personal remarks towards a respected member who hasn't even been in this topic.
By the way, you still haven't asked a question you didn't already answer wrongly and in your egotistical manner tried to slap everyone else in the face with as being right.
_________________ Brian Altenhofel
You spend your whole life trying to remove feedback, and then when you want it, it fights back!
Jp22 Recording Org Pro Audio Forums
Joined: Jul 10, 2005
Posts: 196
Location: Minnesota, USA
I could easily give back any minimal dynamics lost compressing with a bit more eq on my mixer
That says it all. That was the first sign you didn't know what you were talking about.
Oh yeah? Meaning, I can easily bring up the highs (HF's or MF's) on my mixer's eq and once again bring back or hear what I lost dynamically, its simple, if I take a little out at the beginning of the chain with compression I can cerntainly just as easily put it back towards the end of the chain. Shows how much you know!
I could easily give back any minimal dynamics lost compressing with a bit more eq on my mixer
That says it all. That was the first sign you didn't know what you were talking about.
Oh yeah? Meaning, I can easily bring up the highs (HF's or MF's) on my mixer's eq and once again bring back or hear what I lost dynamically, its simple, if I take a little out at the beginning of the chain with compression I can cerntainly just as easily put it back towards the end of the chain. Shows how much you know!
Oh yeah that clears it up. You're an idiot.
BrianAltenhofel Recording Org Pro Audio Forums
Joined: Apr 08, 2005
Posts: 378
Location: Clinton, OK USA
That reminds me of the time I cleared up the vocal track by running the drums through a harmonizer. or something.
Sarcasm?
Here's a good explanation on EQ:
Ethan Winer wrote:
In the beginning all equalizers were analog electronic circuits using capacitors and inductors. These components shift the phase of AC signals passing through them. If you combine a signal with a phase shifted version of itself (after passing through the capacitor or inductor), the frequency response is altered. As one cycle of the wave is rising, the shifted version is falling, or perhaps it hasn't yet risen as high. So when the two are combined they partially cancel at some frequencies only thus creating a non-flat frequency response. Therefore analog equalizers work by intentionally shifting phase, and then combining the original signal with the shifted version. In fact, without phase shift they would not work at all!
Most digital equalizers mimic the behavior of analog equalizers, but with a completely different circuit design. Instead of using capacitors and inductors to shift phase, they use taps on a digital delay line. A digital delay line is a series of memory locations that the numbers representing digitized audio pass through. The first number that arrives is stored in Address 0. Then, at the next clock cycle (44,100 times per second for a 44.1 KHz. sample rate) the number in Address 0 is shifted into Address 1, and the next incoming sample is stored at Address 0. As more numbers enter the input they are shifted through each memory location in turn, until they eventually arrive at the output. This is the basis for a digital delay, and you can alter the delay time by changing the total number of addresses each number passes through or the sample rate or both. (A series of memory addresses used for this purpose is sometimes called a shift register because of the way the numbers are shifted through them.)
To create an equalizer from a digital delay line you tap into one of the intermediary memory addresses and feed a varying amount back to the input. Just like the feedback control on a tape recorder-based delay like an old EchoPlex. Except without all the wow and flutter. You can also reverse the polarity of the tapped signal before sending it back to the input to get either cut or boost. The bottom line is the delayed sound combines with the input - just like a flanger effect - to create peaks and dips in the frequency response. By controlling which addresses along the delay route you tap into, and how much of the tapped signal is fed back into the input and with which polarity, you create an equalizer.
_________________ Brian Altenhofel
You spend your whole life trying to remove feedback, and then when you want it, it fights back!
Jp22 Recording Org Pro Audio Forums
Joined: Jul 10, 2005
Posts: 196
Location: Minnesota, USA
There's the entire quote that you misquoted. It states that no matter what guitar distortion you are getting (tube or solid-state) it is naturally being compressed. That's basic audiophysics.
READ AND UNDERSTAND: I DON'T CARE and thats NOT what i'm here for, are you that hard headed or what? jeeeeezus....
Quote:
By the way, your third post starts the argument.
WHAT ARGUMENT?!?! Who the hell is arguing!?! You guys are all arguing with yourselves!!!!
Quote:
It shows your ignorance and arrogance and how you ask a question and then already seem to have to answer. It first shows up in your second post, but is very apparent in your third.
*yawn* My ignorance and arrogance.... great. Did you even bother to read my first few posts at the beginning of this topic yet?
Quote:
By the way, you've been recycling for a while now, and now your just taking yourself to a whole new low.
Actually, you are! THIS IS MY TOPIC! So get out unless you have something to contribute!
Quote:
Yamaha NS-10's are a standard because of how accurate they are. And then you're making personal remarks towards a respected member who hasn't even been in this topic.
Same old rhetoric. Duh... jeee George, I use monitors I built out of tree sap and the special Japanese bark from a tree I imported from Japan to my backyard, the tree is the same used to make the unwise wise in listening grasshopper for the market kung fu hoes wizard monitors know the best way Danielsson.
BrianAltenhofel Recording Org Pro Audio Forums
Joined: Apr 08, 2005
Posts: 378
Location: Clinton, OK USA
Well, i've had enough. Thanks for ABSOLUTELY NOTHING to everyone who has posted in my topic. You've all been ABSOLUTELY ZERO help. I'll check back one final time tomorrow to read some more of your typical insults and rhetorical ramblings from the confused bowels of insanity itself. Hopefully i'll find a post from someone wise enough to understand my situation.... but I doubt it. Have a nice life AND MAY THE GODS CURSE YOU ALL IF YOU DON'T FORK SOMETHING OVER!!! READ MY FIRST TWO POSTS, THIS IS MY TOPIC!
Last edited by Jp22 on Wed Aug 03, 2005 8:01 pm; edited 2 times in total
Jp22 Recording Org Pro Audio Forums
Joined: Jul 10, 2005
Posts: 196
Location: Minnesota, USA
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