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Thu 28 Feb 08 @ 9:10 am

i think the best advice is to first think and analyze how you want to recall your music. how do you spin? how do you arrange your sets? and most imporatntly what aspects do you remember about music?

some people remember years and labels and whatnot. i organize my music by feeling, so i can move up, down and laterally in my set based on the mood.

any kind of organization scheme is dependant on the selected axiom. the success of that organization structure is how long that axiom works until broken.

to explan further: an axiom in my case is mood. I start with the type: house. subfolders are deep, funky, jazzy, tech, etc. but then in each of those i repeat the same folders. for instance, i have some deep house that is quite techy so its in House > Deep > Tech as opposed to a Funky track which is more upbeat in House > Funk > Up

it works for me becuase i have 1000s of a very similar genre. i dont use or believe in playlist because every time you spin, you need to make up the set based on too many factors to just repeat something done before.

so that is the axiom. i also demonstrated how i expand. for instance, my house > funk folder grew to about 300 before it became too unweildy to find anything. so i spawned the subfolders which are similar in other (i had done in Deep first because i have several hundred more tracks there). so the axiomatic approach is still the same but it allows me to expand.

when you start to mix and match axioms, the house of cards falls down. it becomes difficult to find something.

so whatever you do, think about 'how you will use it' before deciding how it will work. once you figure out how you want to recall tracks, build a solid system and stick with it.

and remember the last point. rework in anything is always more time consuming than doing it right the first time. dropping 2000 tracks in one folder and then deciding to organize is worse than doing 30 everytime your purchase.

Thu 28 Feb 08 @ 8:52 am

its not necessarily about 'hearing' a difference. its about the operations done on the material. for instance, you might not hear a difference between 192 and 320 file until you apply a keylock and pitch-shift it. the differences will become more apparent.

similarly, the differences will become more apparent the more you slow a track down from it's original speed.

a good analogy is this. youre driving on the highway (Rather someone is driving and you are in the passenger seat). You are staring at the dotted lines in the road. Each line segment represents a bit in the mp3 file. as you speed up, the bits are appearing more quickly. if you go fast enough, in fact the line starts to appear like one long line. when you slow down though, the lines come more slow. if this is the audio, the lines (clarity) become so far apart, you loose definition of the track and start to hear artifacts in the sound. often it can be described like a soft choppy sound.

ideally this isnt too much of a factor so the argument becomes moot. but in circumstances it applies. i listen to mp3s of any calibre but prefer 320 if i can help it. however, i refuse to spin with anything less than 320 for sake of quality.


final point. YES, you can hear the difference between 192 and 320, or even 256 and 320.... BUT, you only hear the difference when listening to very good speakers. for instance, my genelecs will show you every flaw in every track. but you will rarely hear this in a club or on headphones because the reproduction and stereo imaging is not accurate enough.

Sun 17 Feb 08 @ 2:15 am

This is from two of my posts in the forum.

A digital signal should not be boosted beyond 0db unless you are using a professional interface that has a programmed headroom beyond. If I remember correctly, commercial products use -4db headroom and pro uses +6dB headroom. If your gear doesnt have a specific setting for this, it means its -4dB. If you ever experience distortion, you are boosting too much. If you dont hear distortion, that doesnt mean it isnt distorted, it could mean there are saturation filters in the circuitry.

The best way to ensure best possible sound quality is stay below 0dB and never boost the EQ beyond 0dB. Use the gain on the last point before going to the speakers (the mixer if you use one).
sound is an analog function. when you push the levels, you eventually get distortion, but before that you get a natural effect from analog circuits called 'saturation' which makes it a warmer sound.

you dont get this in the digital world. it _just_ clips. having said that, alot of digital circuit manufacturers are putting elements in which mimic saturation. but there is a limit to it.

the point is moot when were talking mixers. on my A&H I usually have it close to redline on the channel input because there is a very annoying attribute to my xone:3d when using the digital master output back into my pc for recording..... it's very low. since the digital output is pre-master but post-fader, your signal strength is quite weak. i find this very irritating especially for broadcasting unless i would use the analog output which is post-master and re-digitize it on the way back into the pc. this kinda defeats the purpose for having AD/DA converters in the mixer itself. no matter though, i will be switching to an RME Hammerfall card in the next month or so.

regarding the bit about the pro interface, im talking about the soundcard (whether it is in the mixer or not, like the xone:3d). the whole deal is about signal to noise ratios. if you can get a louder signal out of an instrument and into a recording bus (mixer) without distorting, that means your noise level is drowned out much further than commercial products. this is only for the benefit of recording. once recorded and packaged (cd, download, vinyl, etc) the playback headroom is irrelevant because it's only reproducing the packaged source. if you reproduce a source which had alot of noise, you will hear it. if your recording had a very high SNR then the noise is much less and the percieved quality is much greater.

thats basically how it works. now how this relates to VDJ: Ripped CDs and downloads are _already_ mastered to it's limit. Vinyl is different due to the PVC medium, but I wont bother explaining that. Basically if you boost the gain of something which is already as loud as it can possibly get in the digital domain, it is garunteed to clip.

this is a substantial thing which most VDJ users dont know or realize. you cannot boost a digital signal father than it's maximum headroom. the whole point of CD and Digital mastering is to achieve exactly that. So keeping it in VDJ at -3dB or lower means you are not quite as loud, but you have room for manipulating the sound with effects and EQ without distorting.

users who have an external mixer just need to make the channel input stay out of the red, but by that point, its not digital anymore and saturation comes into play.

for users who use hercules and whatnot, they will very easily distort their signal when they dont adhere to the rules because all effects and eqing is still done in the digital domain.

hope this explanation helps.

Mon 04 Feb 08 @ 12:15 pm

A dude approached me by PM to chat abit about removing vocals. It's not an easy task, so I thought I'd start this thread to show what I know and to start an exchange. Some precursor knowledge, I have DJ'd about 17 years and produced for 10 on and off. Not trained in it, but have access to alot of studios and know alot of people who do it professionally. The following is some data I've picked up on the subject over the years:

Well vocal separation is no easy task. Quite often the remixes you hear (if its a good one) the vocal track was supplied by the artist.

If you need to remove the vocal, this is a whole different story and many producers have their own techniques.

Just to say immediately, you cannot remove vocals on the fly, it's an arduous process you need to do before hand.

The easiest and quickest way is to use filters and paragraphic eqs. Vocals generally tend to be in the range of 1k-5khz in frequency, so you can use two filters to completely drop the lows and the highs out.

The you use a paragraphic eq to 'notch' out certain frequencies of instruments that exist in the same range of the vocal.

There is one other method which I dont use much, but sometimes can be effective, which is to do the reverse of above. You use only a paragraphic eq to remove the vocal and not the instruments, since you getting rid of less portions of the track (only one instrument; the voice). After this, you take the result and invert the wave. Then you mix down the inverted wave over the source. If this is done properly, you will in theory cancel out the instruments and leave a cleaner result of the vocal which you can clean up with some more filters and paragraphic eqs.

Long story short, its a trial and error process that you need to mess about with for a long time. The result is also relative to the source material, as in the quality of result depends on the instruments and composition of the source. Saxaphones tend to compete in the vocal range, and other instruments may. It all depends on the track you're trying to remove.


Sorry, this isnt really a straight answer, because their isnt one. These kind of techniques require alot of post-production and require some knowledge of sound. But you can learn yourself if you spend the time to do it. I suggest having the correct tools as well. Doing it in vdj wont cut it. You need a proper editting tool like Steinberg WaveLab or Sony SoundForge (used to be by Sonic Foundry). With that, you need something like the Waves Gold Bundle VST/DX plugins which come with a quality paragraphic eq.

Essentially a paragraphic EQ allows you to manipulate multiple bands and specify which frequency to modify and how sharp, like taking a big or small slice of a pie. Only with this tool can you zoom in on the correct range. AFter that you can use the onboard spectral analyzer to visually see (before and after the eq) what frequencies the vocals exist in so your work is a bit less trial and error....

Fri 11 Jan 08 @ 6:39 am

Hey all,

I've been mucking about with my Xone 3D for about a month now and I simply love it's control surface. Soundcard leaves more to be desired, but its good enough to just take the lappie and mixer and plugin!

Anyhow, I've spent quite some time working on a midimap which suits my preferences and thought i should share with the community.

This is version 1 of the midimap and comes complete with the template overlays. You can download both the xml file and 1:1 overlay image here:

http://www.djpaz.net/images/Xone3D_Overlay_v1.zip

The labels in blue are for pushbuttons and the labels in green are the knobs. Hopefully some of you will find this usefull. Because there are so many control surfaces and so many options in VDJ, I mapped the ones that I find usefull in my mixing adventures.

NOTES (Wishes):
- the loop select should allow a value or increment so you can use a knob or button to increment the values instead of the stupid popup which is useless if you dont want to lunge for your mouse
- the load_sample function should have a '0' value which allows you to load the currently select sample on deck X. this would allow me to cut the number of buttons down to just one knob for select and one knob for load...