Re: Powered Subs...on to mastering

["Krispen Hartung" <khartung@cableone.net>]

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
>