quote:

Originally posted by cronodevir No storyteller, no one here can even hold up against me in a debate, and I haven't even broken a sweat yet. My superiority is pretty much undeniable.

Yeah you are right, I surrender!

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http://soundcloud.com/subtara

LMAO

/kaput

2am. I am laughing. Hard. .

Back to work.

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Storyteller Website | Storyteller @ Facebook | Storyteller @ Beatport | Storyteller @ Soundcloud | Stephen J. Kroos - Europa (Storyteller Remix)
Anthony Mea - Get It On (Storyteller Remix)

quote:

If less is more think about how much more more would be. -Frasier

First off about the 4050 6 db roll off. This isn't to fix a sound technically in the mix this is to CHANGE a sound and give it a warmer tone. On some tracks this works great...

IF THIS DOESNT WORK FOR YOU DON'T DO IT THEN.
I could try to remember where I heard this advice.
But unless I post a full 3 page interview you will probably say that I ''quote him out of contest too'' There is no winning with some people.


Now about the limiting;

Everytime your track gets played out in clubs it gets processed though various stages of limiting.

Everytime your track gets played on the radio or on tv it gets processed by a limiter only this time the fader is pulled all the way down this is what ''brickwall limiting'' means. As loud as it gets. Coz I heard you toss around that term. (by the way since english unlike flaming is not one of your strengths) I AM NOT ADVISING ANYBODY TO BRICKWALL LIMIT THEIR MASTER CHANNEL YOU IDIOT.

quote:

Originally posted by DJ RANN Existo - could I suggest that this be a work in progress and maybe you could amend it and update as people chip in their suggestions. Stuff like this can be useful for a lot of people, having it for reference and possibly making it sticky worthy?

Anybody reading this can take some of these tips fire up one of their tracks and apply these tips to their music and see if something works or if something doesn't work for themselves.

These are tips I picked up through the year when hearing other audio engineers and producers talk about mixing.

And talking these tips and applying them to my music has made stuff sound better. But If every time I post on here there are 20 mother****** talking about how I am doing it all wrong and how they know what’s best without actually sharing it with the rest of us and contributing nothing but snobbery to the forum they can suck my d*** as far as another tutorial goes.

Last edited by Existo22 on May-15-2009 at 14:24

quote:

Originally posted by Existo22 But If every time I post on here there are 20 mother****** talking about how I am doing it all wrong and how they know whats best without actully sharing it with the rest of us

That is DJ RANN. Telling others they are wrong while contributing absolutely nothing. The way I deal with him is by setting him off, he too easy. He is a troll, everyone knows it, just play along.

The rest tell people they are wrong alot and contribute only half the time.

Diginut, RichieV, Storyteller, and Theran however do know what they are talking about when it comes to production. Id take their advice if they give it...as for every body else, they are neither here or there.

___________________
i'm the alchemist without the cyst without a doubt out of the mist
dig out the grout expose the pest to take it out without the rest

they will attest my patients restorations from the best

i'm from the west i'm not an acmeist only the alchemist

rofl rofl

Do you see, now?







This is why we can't have ANYTHING nice.

quote:

Originally posted by orTofønChiLd basically a sound card does turn an analogue signal into digital is true. People who only use software won't need it.

quote:

Originally posted by cronodevir Hey, someone comes out of the bog with something smart. And its my contention that most people use software pretty much exclusively.

cronodevir and orTofønChiLd, there are huge holes in your knowledge. before you start arguing a point, it usually helps if you actually know what you are talking about. otherwise, you just come off looking like retards (like what has happened in every thread you post in).

the main reason people buy soundcards is for the better digital to audio converters. your onboard soundcard would have piss poor DAC, in the range of ~90db SNR, with very high jitter. my echo audiofire has 113db SNR, and very low jitter. every 3db is DOUBLE the percieved loudness. hence, my output is over 700% better, in terms of loudness in SNR. latency has NOTHING to do with DAC. latency is a software buffer. DAC is a digital circuit in hardware. at any latency, my soundcard will STILL be more than 700% better, in terms of SNR loudness.

why is this important? YOU NEED ACCURATE OUTPUT TO BE ABLE TO HEAR ALL THE DETAILS IN YOUR SONG. your onboard soundcard does not give you accurate output, because it is a piece of shit. you need accurate output REGARDLESS of whether you are using hardware OR SOFTWARE.

another reason why your onboard soundcard sucks is because it has shit quality opamps, which amplify the signal. professional soundcards have high quality opamps and only ever output at line level. likewise, since your soundcard is part of your motherboard, the DAC is not isolated from interference from other things inside your computer. your CPU fan, hard drives and even just crosstalk from circuit tracks that are too close, will cause interference. hence, onboard soundcards also have a VERY high amount of distortion, due to interference. even PCI soundcards distance the circuits far away enough to solve this problem. the best solution, however, is obviously to have an external card.

latency has NO effect on sound quality. if your buffer is too low, however, you will get gaps in the audio stream. gaps != lower quality. latency is just a buffer. the buffer exists because your computer cannot output a constant stream, so the computer needs to "read ahead" a certain amount of time, to avoid problems when the CPU spikes and can't provide a constant audio stream. when you run out of buffer, because your CPU can't handle the load, then you get clipping because the audio stream gets gaps in it. having a larger buffer makes absolutely NO difference on sound quality.

quote:

Originally posted by echosystm cronodevir and orTofønChiLd, there are huge holes in your knowledge. before you start arguing a point, it usually helps if you actually know what you are talking about. otherwise, you just come off looking like retards (like what has happened in every thread you post in). the main reason people buy soundcards is for the better digital to audio converters. your onboard soundcard would have piss poor DAC, in the range of ~90db SNR, with very high jitter. my echo audiofire has 113db SNR, and very low jitter. every 3db is DOUBLE the percieved loudness. hence, my output is over 700% better, in terms of loudness in SNR. latency has NOTHING to do with DAC. latency is a software buffer. DAC is a digital circuit in hardware. at any latency, my soundcard will STILL be more than 700% better, in terms of SNR loudness. why is this important? YOU NEED ACCURATE OUTPUT TO BE ABLE TO HEAR ALL THE DETAILS IN YOUR SONG. your onboard soundcard does not give you accurate output, because it is a piece of shit. you need accurate output REGARDLESS of whether you are using hardware OR SOFTWARE. another reason why your onboard soundcard sucks is because it has shit quality opamps, which amplify the signal. professional soundcards have high quality opamps and only ever output at line level. likewise, since your soundcard is part of your motherboard, the DAC is not isolated from interference from other things inside your computer. your CPU fan, hard drives and even just crosstalk from circuit tracks that are too close, will cause interference. hence, onboard soundcards also have a VERY high amount of distortion, due to interference. latency has NO effect on sound quality. if your buffer is too low, however, you will get gaps in the audio stream. gaps != lower quality. latency is just a buffer. the buffer exists because your computer cannot output a constant stream, so the computer needs to "read ahead" a certain amount of time, to avoid problems when the CPU spikes and can't provide a constant audio stream. when you run out of buffer, because your CPU can't handle the load, then you get clipping because the audio stream gets gaps in it. having a larger buffer makes absolutely NO difference on sound quality.

who is this echosytem guy?

Fuck me, you lot argue over some pointless shit.

Everyone is different and everyone has their own idea on how a track should sound.

Music/Audio is not an exact science... it is all very relative... and let's not forget, should you be signed, all the hard work you put into digitally tweaking your track will be probably overdone by the labels sound engineer/mastering engineer

quote:

Originally posted by orTofønChiLd who is this echosytem guy?

It`s a bot.

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