Tanqsicc

Please help! I need some DSP preset for max quality of sound.

Maddogg6

the least processing/changing of an audio signal is generally considered of highest quality. To do that, you need to start a clean DSP, but to do so also means limited functionality for things like 2.0 > 5.1 & 5.1 > 2,0 converting - thus, what a 'musician' needs will dictate what the DSP will look like - I have several configs I load as I need that I made myself...

.. you really should learn how to make your own..
http://www.hardwareheaven.com/effects-...-your-dsp.html

Tanqsicc

I aint understand this guide... when I try to do some config its always have no sound.
Im lookin for 2.0 presets.. 3539 version of KX. and, if it means something, i'm working in rap/r'n'b styles.

Maddogg6

Did you adjust the 'tweak' screens for the plugins you added?
Do you have the 'Swap Front <> Rear' switch set correctly on kX MIXER - Master page?

Is there any step in the guide you didnt understand in particular?

Styles makes little difference - what software you use has more influence on your DSP configs... Are you recording/producing? or playing back with DJ/mixing software and needing a headphone monitor?

kX requires a bit of a learning curve, and while some presets exist that you can download (kX preset exchange), I have found that often ones posted by others wont work due to the various versions of kX, various differences in SB card models and the version of the popular plugins. I typically post pics of the DSP instead of kX config files for this reason.

Tanqsicc

I use Logic and Nuendo for midi and producing. no headphones right now, only 2.0 monitors.
No one preset from "KX preset exchange system" is never works on my system.
sb0400 10k2
About guide - I dont understand everything))) what plugin i need... e.t.c)

Russ

The guide uses mostly ProFx plugins, which can be found here:
http://www.hardwareheaven.com/effects-...-mx6-ac97.html
(make sure you get the latest version, do not use the bundled version of ProFx)

Note that the plugins used in the guide are an older version, so there will be some small differences from what is in the guide (i.e. use ProFx:ADC for AC97 sources instead of ProFx:SRC).

Tanqsicc

Tnx!! It works! It was a stereo mix - gain 0))) In mixer i have no found some faiders, its normal?

Tanqsicc

now my DSP looks like this

Maddogg6

Yes that is normal;

INS & OUTS page TOP row faders relies on Epilog plugin...
INS & OUTS page bottom faders relies on Prolog plugin..
RECORD PAGE Bottom Row faders relies on Epilog Plugin..
RECORD PAGE TOP row faders relies on XRouting plugin..
MASTER PAGE - SPEAKER SETUP SWITCH relies on Surrounder

Using Profx ADC will prevent needing to use kX Mixer AC97 page - its best to only adjust the ProFX ADC if you use that plugin for AC97 sources.

After no longer using those plugins - you dont really need kX mixer - set kX icon to open DSP on a double click instead of the mixer...
(Actually - dbl clicking on epilog, xrouting and prolog opens the kX mixer to their respective pages - so, I change the dbl click action for the kX tray icon anyway, even if I use those plugins)

Russ

Do not use "Stereo Mix + Gain", it has a bug.

That config uses ProFx:SRC set for Line In (AC97). If you want to use that input, you should use ProFx:ADC instead (as I indicated in my previous reply).

As for the missing sliders in kxmixer, that is normal, as those slider are associated with plugins that you do not have loaded. You can control some aspects of what those sliders would do, using the tweak windows for the various plugins, but some functionality may be missing when using custom DSP configs (i.e. you will not be able to control Master Volume, etc with the above config).

<edit>oops Maddogg6 beat me to it... </edit>

Tanqsicc

Ok, now I use mx6 (on place of stereo mix) and ADC on input - Line In AC97..
How this DSP settings influence on quality of sound?

Lex Nahumury

That depends on your definition of 'sound quality' but as mentioned already, the 'best' sound path through DSP would be the shortest one.
(not even using any DSP mixer plugins)
i.e;
prolog -> epilog (for rec),
fxbus -> epilog for playback

which is roughly the same as;

ADC -> ASIO (for rec)
SRC -> KXLT (for playback)

But frankly I'm 'amused' by your concern about sound quality *in* the DSP path while you record through AC97 which is a very poor device in terms of 'sound quality'.
Remember; Garbage in = garbage out!
So I wouldn't worry to much if I were you

cheers.

Maddogg6

to more directly answer this question - a common problem with 'sound quality' stems from improper gain stage settings - thus, the fewer there are, the less chance of inadvertently re-amplifying an attenuated signal, and thus amplifying noisey signal...
So - for optimal sound - (optimal gain stage settings)

1) make sure you have a strong, but never overdriving signal comming in on Line1
2) Putting a PEAK meter at all stages to ensure optimal levels and use MX6 to adjust volume to taste - never changing ADC levels until you change the source feeding it - even then, try to adjust the source its self - to again, provide a max volume with out overdriving the LINE in's electronics.
If you overdrive the electronics, no matter what you do in the DSP, it wont fix that distortion.

The above addresses 2 major things...

1) Keeping signal to noise at a max - AC97/ADC is rather noisy, so feeding line in a strong level helps hide that fact, and we will limit re amplifying a noisy signal.
2) Simple DSP config means less chance of attenuating > amplifying - which hurts signal to noise ratios. And, the least processing on a signal is best for reproducing what was recorded. Unless those processes are compensating for system short commings - EQ to tune speakers/room as a for instance. But a system without the shortcommings (and CL cards, consumer speakers have plenty) is even better. So if you are an audio purist - you bought the wrong sound card.

If you are just trying to get rid of a noisy LINE in - look at the source and follow the steps above for optimal gain stage settings.

Tanqsicc

Ok, tnx!

WxMan1

What he said.

As nearest as my neophyte knowlege will get me, and plenty of analog recording experience (with 2-track stereo on 4-track tape medium), SNR is important, but so is dynamic range.

All electronic components introduce some noise into a signal (as does ALL amplification).

Attenuation applied to a signal will alter the amplitude of the major waveforms within a signal that will result in loss of information that can not be recovered easily through amplification without noise filtering. The information is still there, its just that its present for less time (area under the curve). Increasing the area under the curve - amplifying the amplitude - will introduce noise. Physics knows of no quid pro bono phenomenon.

It is far better to allow minimal clipping (distortion) of the signal, i.e., <10 ms, rather than erode the dynamic range of a signal through attenuation. Fidelity of a relatively loud signal (containing weak transients) can never be recovered if attenuation has been indiscriminantely applied to it. To avoid clipping one must sacrifice DR, but to mitigate loss of DR one must accept some clipping in high DR passages.

Nevertheless, and that notwithstanding, ALL levels - hardware and software - in the signal path should ideally be set to near nominal (extremes should be zealously avoided); 5-stage series of 5% attenuations is far better than one 27.6% attenuation (both mathematically equivalent).

Russ

The other point to be made is that music is usually mastered in the way that it is intended to be heard (without any filtering), so anything that you do that modifies the original audio, takes away from the intended experience. Of course not everyone has perfect sound systems, so filtering is used to try and make up for any short comings that a system might have (or to suit personal preferences), but filtering is not likely going to improve the quality (it may sound better to you, on your speaker system after filtering, but there is no one perfect config that will work for everyone).