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Looper's Delight
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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 >