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Hey folks

OK something is bothering me. Lately I've been working on a track, using Cubase, pulsar2, A16 Ultra. All sounds coming from my Virus B synth and Roland XV-3080 sound module.

I am learning compression, limiters and other things, as I want to make my overall sound more professional.

For some reason, even though I've added gain to each channel independently, added compressions where needed (to my ear), and an expander module on the overall mix... it is still quiet.

By quiet, I mean that when I played the mixdown on my consumer hi-fi, I had to crank the volume up much higher than if I played other CDs.

What is interesting is that on the VU in both Pulsar, Cubase and Sound Forge, the level is quite loud, going all the way to -1dB (on Kicks and loud basses)... But really it is not loud.

So what do you think is the problem? Would you recommend a particular book going through a typical recording and mixing process step by step?

Thanks in advance to all of you guys. Please reply.

I've thought of using some kind of "maximizer", but isn't what pulsar's "Stereo Expander" is?

-mano

<font size=-1>[ This Message was edited by: mano1.com on 2002-05-06 07:57 ]</font>

An expander is NOT the same thing as a "maximizer".
An expander is actually the opposite!!!
It is like a softer version of a noise gate.

A "maximizer" is a combination of a multiband compressor and a brickwall limiter.

So what does that mean?

A multiband compressor splits the signal into different freq.bands and applies compression separatley to each freq band.
This allows a much smoother compression on a final mix as the freq bands don't interact.
A bass drum does'nt lower the volume on a hihat or cymbal.

A brickwall limiter is limiter where there is absolutley no overshooting. Not a sigle sample over the threshold.
The only way to achive this is with a digital limiter.

If you use these together comp. first limiter last you can push the levels up so high that the audio file looks like a squre block.

Do you wan't to? (does it really sound good?)

Use your ears.

A limiter to recommend is waves L1. It is absolutley fantastic.
My favourite multiband compressor is DUY Shape although most people use the Waves C4 i guess. It has been getting rave rewievs.

You say the file "looks" loud but isn´t....

The reason for that phenomenon is quite simply that you have too much "low frequency energy" in your mix. --> Use a high-pass filter at 31-40hz. You are on the save side when you stay under 35hz. Compression in the mastering stage is only used for controlling things rather to gain loudness, but of course it´s possible to gain loudness whith a compressor when you set it up as a limiter.... Just using a limiter whithout watching the waveform is dangerous because your ear gets accustomed to the dynamics and you might loose bits by killing the dynamic range. So first you have to optimize your frequency balance of your mix THEN apply some dynamic tools like "compressors" or "limiters".

When treating your finals keep in mind that there are different versions (CD, radio, single). When you compare your mix to the stuff you hear on the radio, it´s better to have less volume than those mixes. Compare your stuff to mixes of the same genre (CD version)!

Another thing that is important for perceiving loudness, is that the human ear is more "sensitive" to amplitudes at certain frequency ranges. That´s the reason why radio mixes tend to have more energie at 2-4,5khz. You could just as well boost those areas whith a eq and use that phenomenon to your advantage.

Out of tune instruments and the "intrument balance" can also lead to a loss of loudness.
In the case of too much "bass" and "bassdrum" playing at the same time your frequency balance does suffer by having too much low energy again. So the main reason for your problem is definitely the load of subsonic frequencies or other sounds that do not compliment the music, but rob the space!


Regards,
Sunshine

the previous posts are all good info.optimaster is a nice multiband comp. remember,finished cd's go thru an intense eq and dynamics session known as "mastering" b4 you hear them.just get your best mix.(don't over compress) and then finish it later when you have a bunch of stuff to process.another good thing would be to hire a professional mastering session.watch, listen, ask and learn. a good engineer often likes to discuss his work and he can tell you what probs he hears,often helpful since he hears many mixes.
if you must be a do it yourselfer,you must experiment.listen to stuff that sounds good and try to emulate it.screw around with it.you can't break it.remember it's eq and dynamics that = loudness not just volume.contrast helps.to much low end doesn't( eats up all the amps power on freq the speakers can't reproduce)


<font size=-1>[ This Message was edited by: garyb on 2002-05-06 10:36 ]</font>

Mano I assume you have your A16 Ultra dips set for -10db signals?

If you want to get some good final compression on your mix to make it as loud as possible. Try out the free Endorphin compressor from http://www.digitalfishphones.com
It's a dualband compressor and analog saturation emulator.
This plugin is really good quality and if you push it it can really push your volume up loud. (although it's best to treat it a bit more gently to prevent serious pumping and breathing in your mix).

You say that on te VU meters the level is loud but it doesn't seem as loud as normal "professional" recordings. This is correct because actually the peak levels are at the same volume, but if you don't use compression, there will be a greater dynamic range than on a mastered and compressed recording. In other words, there will be loud peaks, but there are also quiet parts, and there is a large difference between the quiet "peaks" and the loud peaks. You can see this on your VU meter because it goes moves around a lot.
What you see when you use Endorphin (or any other mastering compressor), especially when you use it at high settings, is that the range in which the VU meters move has gotten smaller. So the peaks are actually at the same level, but the queit parts are boosted so they are also at a higher level. There is less difference in volume between the quiet and the loud parts, the sounds is COMPRESSED. This also means that the sound has become denser, and even though the peak volume is the same, the overall power of the sound is much bigger and perceived louder.
This is basically what compression does.
If you try out that plugin you'll see (and hear) immediately what I mean.

WOW guys great posts! THAT's what I call help. Thank you so much!

subhuman: funny, I didnt know there were dip switches on my A16U! I just looked and saw something looking like it in the back. I will experiment with it. Its on the default setting, do you happen to know what it is? I noticed that even at full volume, the meters werent lit more than 2 LEDs up.

I guess the switches are attenuators, to adjust for high/low level inputs. Set them so you get as high volume as possible, without clips.

Another thing you could take into account is that, for best signal-to-noise ratio, you have to send as loud as possible over your (analog) wires, also without clipping/crossing the red line of course. Look at it this way: the noise on a cable will be about constant. So if you raise the 'used signal' level, you get less noise.

One thing I always do, is to put a 20-30Hz highpass or lowcut filter in front of a compressor. These low frequencies take lots of energy, while they're inaudible. It allows for the compressor to raise the other frequencies more.

If you have a sidechain, you could put the lowcut in the sidechain, but that's not advisable for master compressors. In mastering, you'd want to go as close as possible to 0dB (for dance/club, that is). So only use it for 'intermediate' or channel compression. The low frequencies will still remain in the signal, while they do not attenuate the higher frequencies. You can remove the subs in the mastering only.

Lowpass filters are also handy for other reasons, ie. with one device I once created DC signals! This is possible in the digital domain, but you'd never want DC to go to your speakers. I noticed it when I had multi track recorded a song. Luckily, my analog mixer has condensors which filter the DC, so my speakers remained unharmed. But my phones were connected directly to one of Pulsar's analog outs, and I got a little strange feeling during recording...

One developper once told me he was thinking about making an airco synth which would produce <10Hz sounds. Don't try that at home!

Of course it´s advisable to cut out all unnecassary frequencies, whether they are at the bottom or at high regions (in your mix!!). All this cutting is done to get rid of frequencies you can't hear that are eating up headroom...


For your "finals", it basically all goes down to what you have done while mixing! When you apply a "high-pass" filter and the sound of your mix changes more radically you have probably not done any low cut on your single tracks...


I don´t think that it´s so common anymore to use "LPF" at the mastering stage. In the vinyl days "LPFs" where more necessary than today. There was simply no reason for driving these high frequencies that eat amplifier headroom and add distortion....


For "CD productions" there are a lot of "harmonics" up there that put an air around the music. "Low-end" is the bigger problem. I would only use "LPF" when it is sonically necessary, and you have no access to those single tracks/groups anymore...


Keep in mind that there are different "pass-filters" out there that do have different characters and different slopes. "18db" or "24db"/oct slope set at 30 Hz shouldn't do anything to the midrange and will tighten up the sound.


Regards,
Sunshine

On 2002-05-06 10:42, subhuman wrote:
Mano I assume you have your A16 Ultra dips set for -10db signals?

I just did that and now I got a huge noise coming from the A16U I posted a new thread about this problem

Thanks for all the good posts guys, this week-end I am going to look more into it.

-mano