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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
>
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