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Re: Powered Subs...on to mastering



dear krispen,
 
i think commercial mastering is about compression. especially multiband compression. i use multiband compression as a dynamic eq for mastering rather than compressing to get more db out of the piece.
 
i think first insert a high pass filter which will cut below 30hz. then the typical waves mastering eq, multiband comp, limiter setting. you should be careful about maxbass. i think you should use that plugin mostly for subbass oriented dance, electronica tracks, it may really destroy a proper rock/pop mix.
 
for waves L2 you should not go below -5 db. after that it starts to degrade the signal. ( to my ears )
 
set the multiband compression so that it acts as a dynamic eq. you have to control the peaks in frequencies and for that it is better to use multiband comp than a eq. i use all the 5 bands of the waves multiband. you need to experiment with the crossover frequencies of course. you can compress 4-5 dbs of the sub bass and bass range. this will give you a much more punchy and clear presentation of the piece.
 
also if you can, try to invest money for an analog mastering eq. i use manley eq for my mastering sessions. even a 1 db boost at 80 hz gives you a such a loving, controllable bass.
 
best.
 
----- Original Message -----
Sent: Monday, January 07, 2008 5:26 PM
Subject: Re: Powered Subs...on to mastering

I've been doing a lot of mastering and mixing lately on a project and have
learned a lot of new methods and techniques.  I've heard folks say mastering
and mixing is a black art, now I know why. In these particular songs, they
sounded wonderful on my headphones. There were some really cool and deep
things going on in the 44hz range and below, and some others in the 62hz
range. It all sounded great through my headphones, but those frequencies
were reeking havoc on my consumer stereo systems - car stereo, portable
stereo, etc. They were really prominent resonant frequencies that were
rattling the hell out of the speakers and causing distortion.  And it wasn't
a level problem...all my stuff was compressed/limited and below 0db, and
there was no redlining in my original recordings. It only had to address
troublesome resonant frequencies.  So, I had to go back and re-master the
files, adding a high pass filter that rolled everything off below 60hz. That
did the trick, but I really miss the sound in the headphones. And I'm sure
there are some hi fi systems that would have produced the original files
well, but I can't expect everyone to have a system like that.  Then I
started fine tuning some of the other songs, doing a frequency spectrum
analysis, watching and listening for other resonant frequencies, unusual
spikes, etc....correcting them with various parametric EQs and so on.  Then
it got complicated, because if I was altering a whole mix, then I could not
fix one problem from an instrument in the mix, without changing the
frequency of another instrument...so I go back to the source tracks/wavs,
etc, etc. I could spend hours and hour just on one song and still not be
satisfied with the results, or waver between two different approaches.   Is
there a simpler approach?

I'm wondering what others uses as a consistent approach to mixing/mastering
their music.  For example, after you remove the DC offset, do you apply a
unique approach to applying EQ? What about compression/limiting?  On
average, how much of a threshold do you apply? Do you suck the dynamic range
out of your mixes to maximize volume, or are you very conservative and
preserve as much of the original dynamic range as possible, sacrificing some
volume. What sort of tools are you using? I use Waves L2, and the whole
sweet of others in that package.  Ever use Waves MaxxBass? I read some
articles that recommended it during the master process, but I did not like
the results. It altered too many other frequencies in my mix beyond my
original intent.

Moreover, the idealist/purist in me would like to preserve as much of my
original dynamic range and frequency character as possible.  And, quite
honestly, if I ever catch a sound guy altering the EQ on my guitar when it
is was not meant to correct a problem but only server his own idea of how a
guitar should sound, he will hear some sharp words from me.  I spend a lot
of time on the tone of my guitar, and do not appreciate a sound guy
butchering it because of his own sound aesthetic.  As they say, "If it ain't
broke, don't fix it."

So, if I want to preserve as much of my dynamic range and EQ as possible,
what is the bare minimum I should be doing to my final mixes to ensure they
don't generate problems on the average listener's stereo system?  One source
I found said to elminate anything below 60hz because most systems wouldn't
be ableto represent it.  I suppose if I wanted to be a purist, I would only
ensure my overall level is at or close to 0db, and not apply any compression
whatsoever...because once you do that, you are already altering the original
dynamic range of the piece. Then, in principle, I should not have to mess
with frequencies with EQ whatsoever, unless there are serious playback
issues on common stereo systems. That is the direction I would like to head,
but I struggle with competing with other mixes out there in the same genre
that are so ridiculously loud because of the amount of compression/limiting
applied, followed by level increases.  How much of a change in dynamic
range, from original source to mastered recording can a human ear identify?
If, just as an example,  I start with a -60db to 0db range (where only 10%
of my material is above -10db), and master my file so that 40% of my
material is above -10db, what am I sacrificing to obtain an overall
perceived increase in level? I suppose this is where the black art comes in,
because it's not as if there were a low of physics that dictates how this
should be done; rather it is based on subjective or relative engineering
practices.

Any thoughts or best practices would be appreciated here on how to be both a
sound source preservationist, yet a playback friendly sound engineer at the
same time.

Kris




> Krispen Hartung wrote:
>> As many folks know on the list, I use laptop processing via max (looper,
>> other octave effects) that completely transform the sound of my guitar.
>> It is not uncommon for me to play a low E on the guitar (82.4hz), and
>> then apply a two octave drop.  I'm not sure what that would be.
> Divide the frequency by two for each octave you drop.  (Multiply by two
> for every octave you raise.)  82.4/4 = 20.6Hz.  You're definitely into the
> subwoofer's range.
>
> Cheers,
>
> Bill
>