Yep - if you are on a budget, you will get what you pay for (Behringer)However, there are a number of plugins that will get desired results from Sonic Foundry's own Wave Hammer up to a more expensive option like the Waves L1 or (really cool) L2 mastering plugs. I use Waves and it is awesome...

Cuzin B

Nearly everyone out there uses the Waves L1 plug-in or L2 hardware. They're look ahead peak limiters/compressors as well as noise-shapers and dither products. Waves also does some psychoacoustic tweaking to make it sound louder. They're magic bullets to get CDs as loud as the squashed stuff put out by the majors. They also change the sound of your mixes a lot.
bgc

I like them all Wave Hammer, Vintage Warmer,Waves L1,L2 but my pick is L2
You can push it VERY hard and still have a clean sound.
try the demos before you buy.

George Ware

Thanks Everybody, I'll check into it and see how it affects things.

John

Bring your stuff to my mastering studio and I'll match those levels right up for ya. I haven't heard the Behringer mastering device, but my T.C. Finalyzer sounds great when used correctly.

Rednroll

Izopte Ozone is also worth looking at-

SonicEPM - Could you ellaborate on "Izopte Ozone" a bit?

Also, for using the Wave Hammer, what are the logistics?

1. Put the effect on the original track
2. Render the original track to a new one

???

I have soundforge 5. Should I open the track in SF and add the Wave hammer there?

I usually use Wave Hammer (in Soundforge) to beef up 2 track stereo mixes done in Vegas. Then again, I sometimes throw it onto single tracks during my multitracking in Vegas...just depends what you need at the time.

I have worked with IzoTope Ozone but it really, really alters the mix if you are not careful...changes it to the point of being overbearing (I probably didn't spend enough time with it)

You are probably better off staying with a good flexible preset chain like: Sonic Foundry Compression->Multiband Dynamics (or Paragraphic EQ)->Waves L1(or L2 or Wave Hammer). This makes for a great mastering chain.

My current favorite mastering chain is just 2 plugins - the BBE Sonic Maximizer into the Waves L2 Masters Plugin using the Hi Res CD Master preset)...First - I spend a long, long time getting a premium 2 track stereo mix from Vegas with as little effects as possible (I believe if the multitrack session was tracked carefully, you shouldn't need a ton of sweetening after).

Then I mix down from Vegas and bring that finished 2 track stereo master into Soundforge and apply this chain. It really takes the song to the next level without squeezing the life out of the dynamics of the song itself.

NOTE - I also watch my levels very, very carefully all during the Vegas mix...My finished 2 track stereo mix never peaks past -6db in Soundforge. This is what really helps this chain shine - you must have some decent headroom available when applying this kind of multiband maximizer effect...if you bring in a mix that is already peaking at like -0.2db, an effect like the L2 will just crush the life out of your work...

Cheers,

Cuzin B

Thanks Cuzin B, that helps a ton! I'll try it tonight and we'll see how it goes!

John

Could someone elaborate why it is necessary to keep levels below -6dB (and thus to sacrifice quite a bit of dynamic range, ie. signal-to-noise ratio) if the mix is to be compressed at mastering? I can't see the technical reasons for this. This would also mean that any mix that is to be mastered should not peak above -6dB - is this true?

Jens

Not true at all.
Many mastering engineers will convert back to analog for certain processes anyway making this whole subject moot.
As for leaving " wiggle room " if the guy is staying all digital, hopefully al his gear will have a little more headroom to deal with, before mashing it down to 16 bits.

StateOfGracie,

I know the whole -6 thing can be a little strange to wrap your mind around if you still can't let go of the analog metering world. I know that I was asking myself the very same questions up to about a month ago but since reading all the posts I could find in here on levels, my whole outlook on sound mixing has changed for the better by bringing the overall level down before "mastering" so to speak. I don't profess to be Bob Ludwig or anything but giving your final 2 track master the necessary breathing room (-6) before applying a maximizing plug makes a world of difference.

Try this little test and see for yourself...take a good mix of one of your tracks and mix it as close to digital zero as possible....then take that track and run it though the L1 or L2 (or Wave Hammmer or some other maximizer plugin)and listen carefully to the end result. Do the same with a track that peaks at -6 or -4....to my ears, the track that has some headroom begins to open up and breathe following the mastering phase whereas the track with no headroom left gets maximized again leaving a very harsh ,thick. almost pulverizing sound that doesn't sound natural at all. All the dynamics that may have been present get pounded into the ground because the mix is completed saturated.

This is dependant of course on the style of music you create and the kind of impression you are trying to make. But the -6 rule really works for my material, thanks to some great help from folks in here like RednRoll and others. Just run a quick search in the Vegas Video Forum on "Levels" and you will find a great digital audio learning experience waiting for you.

Cheers,

Cuzin B

Hmmmm, can someone give me a technical explanation as to "Why" this is "not true at all" to leave -6dB of headroom?

Thanks for the plug CuzinB, I believe I gave you the full technical explanation awhile back as to why mixing to levels of -6dB is a good idea. I guess, I'm wondering why working in the digital realm has any difference in this advice when mashing down to 16 bits?

RednRoll,

Your advice on levels was a real eye opener for me and after some careful experimentation, all of my mixes now sound tight, clean and very dynamic by leaving this -6db of breathing room. Prior to this eyeopener, I used to do as many do in this forum - mix with my eyes and mix like I am using analog meters....you know - "Let's get as close as we can to zero" but after applying a Waves L1 or similar to that mix, you can actually see the peak waveforms getting their tops cut off, a sure sign of a good pulverizing. Some of my mix waveforms looked a brick after this thrashing (sounded like it too!)

Now with the -6 rule in place, when I bring a mix into Soundforge and apply my BBE and follow that up carefully with the Waves L2, instead of the peak waveform tops getting cut off, the peaks actually peak properly at -0.2 resulting in a very dynamic sound the you can really get into...I can "hear" the room, I can "hear" the "air" in the mix...hard to describe but I was never able to get this until I backed off the overall mix level before a mastering phase.

The bottom line is - my mixes will have a uniform punch and play well on any system without killing the spirit and dynamic of the song. Good stuff that everyone should try.

Cheers,

Cuzin B

OK - I'm having a little trouble with this -6 db headroom thing....

First off, let's look at an example:
Suppose we have mix 'A' that has a dynamic range of 'R' and is peaking at -0.02dBfs and mix 'B' that has the same dynamic range 'R' and peaks out at -6dBfs. Mix 'A' has an RMS value of 'x' and mix 'B' has an RMS value of 'y' both of which are lower than the desired level 'z' (post-mastered level). Even though their dynamic ranges are identical, mix 'A's RMS level 'x' is higher than mix 'B's RMS level 'y' simply because it's peak levels are higher. (Mix 'A' is already louder, right?)

If we run both mixes thru our mastering chain (for simplicity's sake we'll use only a volume maximizer/limiter), which mix do you think is going to require more processing to achieve the desired goal of RMS level 'z'? Mix 'B' of course! Both mixes will get compressed to some degree, but mix 'B' also requires that the overall level be raised an additional 6dB! The threshold of the limiter will be significantly lower for mix 'B' than for mix 'A' because it has to dig in deeper to pull the material out.

Now, I don't know about you, but I would rather not push my limiter that hard. ANY processing of the original audio signal is going to cause some distortion - in the true meaning of the word - so I think it's to the song's best interest to process it as little as possible.

I do agree that mixing engineers need to leave a little headroom for the mastering engineer simply because if the mastering engineer decides that the song needs to be EQ'd first, he will need a little room at the top for any peaks that might grow - but I'm only talking 2 or 3 dB here, not 6!

The mistake that too many mixing engineers make is they try to make their mixes LOUD. Let the mastering engineer do that. When you strap a compressor across the master buss to try to give your mix some oomph, you are lowering your dynamic range (squashing your mix) & making it more dense. Then you want the mastering engineer to bring it back to life?! If you want to compress, compress the tracks - don't kill them, though. This will tighten up your mix & make it more punchy than compressing the whole mix. The mastering engineer will most likely compress it anyway - and his gear is better suited for that than yours.


As far as the analog vs. digital mastering issue:

If the mastering engineer chooses an all digital signal path (for example: mixes on CD-R digitally transferred into a DAW, processed completely within the DAW using plug-ins & digitally recorded onto a Master CD-R) there is no such thing as "the mastering guy's equipment having additional headroom". Digital '0' is digital '0' no matter what format its in. If your mix on CD-R peaks at -0.05 on the CD-R & it is transferred digitally into a DAW, it is going to peak at -0.05 there too (assuming a Pro sound card was used - if not, who know's what it will do). Analog gear has headroom - good stuff has more than cheap stuff, digital gear has NONE. ANYTHING over 0.00dBfs in the digital world is a flat peak - clipped.

Jeff Lowes
On-Track Recording

Hold on now, if a 16 bit final mix is brought in that peaks at -0.3 dB, yet most of the material is around -9 to -12 db for the most part, doesnt it depnd on where the mastering engineer's digital equipment packs the extra zero's on a 24 or 32 bit system ?
Even in most plugins there is additional bits so that a process can be carried out without clipping, though I guess it must therefore turn the level down if anything went above before it goes out? Im confused here. Are the extra 8 bits always at the bottom ? or are they on top ?

Jeff,

I hear you on the -6db thing...your mix "A" that is real close to digital zero to begin with does require less processing but to my ears, if you apply a maximizer/limiter to that mix, since there is no where to go, what is really happening? To my ears, the mix becomes very dense and very distorted since there is no room to move.

For my preferences, I am striving for the best sounding mix without clipping the tops of the waveforms. I want my peaks to be true at -0.2db with no chopping. If you see a chop, you have exceed your limiter setting and have just had some source material removed.

I also forgot to mention that my multitrack mixes from Vegas have absolutely no compression at all (except maybe a wee bit of taming on the occasional snare hit). When mixed down, this -6 mix is chock full of the original dynamics and sounds really good standing alone on it own. But as soon as I apply some BBE or some Waves L2 - (and I understand that I am "altering" the original mix by doing this), my songs just seem to leap out of the speaker and drip with "air". To me, whatever is getting added in the plugin processing sounds sweet due to the fact I have some room to apply it. When I try this with a hot mix (-0.2db or so), the effect is not pleasant to listen to...too much limiting just wreaks it for me.

Again - this is a very subjective thing and I don't expect it to work for everyone but I really learned a good chunk about dynamics by lowering my levels instead of cranking them.

Cheers,

Cuzin B

>>but after applying a Waves L1 or similar to that mix, you can actually see the peak waveforms getting their tops cut off, a sure sign of a good pulverizing. Some of my mix waveforms looked a brick after this thrashing (sounded like it too!)<<

Actually, a sure sign of a good pulverizing by a poor volume maximizer. I used to experience the same thing until I started using a new maximizer/limiter. Now I can get more level & punch without ANY clipped peaks - and that's without applying this "-6dB rule".


With regard to the bit depth -

The extra bits are for digital word resolution - not for volume. When any given peak reaches 0.00dBfs, it is still 0.00dBfs whether it is 44.1/16, 48/24, 96/24, 32/8 or whatever else.

JL
OTR

Jeff,

Interesting. Would you mind sharing the name of this tool(s) or your technique for your mastering process?

Cheers,

Cuzin B

PSP Vintage Warmer.

BTW - the tool I used to use was the SonicTimeWorks Mastering Compressor. It is basically the same thing as your L1.

I just now tried what you said - I lowered the volume of my track by 6dB and substituted the TWMastering Comp. for the PSP VW, kept the other plugs' settings the same & cranked down on the TWMC's threshold 'til I got the desired output level. After processing the file, I checked the peaks in identical spots in this file vs. the file that was created using the VW. The new file (TWMC) had all the same 'ol table-top "transients", chopped peaks that you describe as "wreaking to your ears" (that's why, you are listening to digital distortion). The file created using the PSP VW has no clipped peaks whatsoever, NONE! That's how you retain that "air" you spoke of.

The equipment used in the mastering process is critical to the outcome of the sound. That is one of the reasons that top mastering house sevices are so expensive (another is experience). I recently checked into some mastering gear from a high-end manufacturer and the list price for their 2-chnl mastering compressor/limiter is $5800!! But you know what?, it's worth every penny of it becuase it is top-notched electronics designed by top-notched engineers. This stuff really does handle the audio signals extremely well. (Uhhh, no, I didn't buy one - just a little out of my budget for this year!)

I was so impressed by the way that the VW handles the transients. That is primarily what sold me on it. No, it's not equal to that $5800 unit, but it comes closer to it than any other plug-in I've ever tried.

Their web site is www.PSPAudioWare.com . Download the demo & try it out for yourself. I think you'll be amazed.

JL
OTR

Jeff,

Thanks for the info. I will check out the Vintage Warmer and let you know what my results are.

Cheers,

Cuzin B

Bizzzzzzzzzzzz.....wrong answer. The extra bits are neither in the top or bottom, they're actually more in the middle. Try taking a 16 bit wave file and normalizing it to 0dB peak in Sound Forge and save it. Now start a new Vegas project with 24 bit resolution and open that file on a track with the fader set at 0dB. Look at your master fader...did you gain extra headroom? NO...it still peaks at 0dB like it should. According to your mastering theory I should have gotten 8 bits more headroom right?

Here's how it works: Think of sampling audio by putting an analog waveform drawn on a piece of graph paper, and assigning it a point using an X/Y axis. Think of your X-axis divisions as the "Sampling Rate", and think of the Y-axis as the bit resolution. Let's say my actual audio is 1 second long and has a digital peak level of 0dB. If my sampling Rate is 44.1Khz, then my grids on my graph paper on the X-axis will have 44,100 points divided into that 1 second of time. If I increase my sampling rate to 48Khz, then I will have 48,000 divisions within that 1 second of time. So on my x-axis the resolution of accuracy increases, as I increase the sampling rate when plotting points that correspond to my original analog signal. Now for my Y-axis, as I mentioned this corresponds to the bit resolution, which actually corresponds to the amplitude of the original analog waveform. So if I have 16 bits, I have 2^16 (65,536) number of divisions representing the amplitude from -Inf. to 0dB. If I have 24 bits I now have 2^24 (16,777,216) number of divisions representing the amplitude from -Inf to 0dB. So basically the grid lines on my Y-axis has increased (ie become closer together)....thus the term "increased resolution" in assigning a numerical value. So the digital number that get's assigned to the sample is more accurate to the actual waveform.

The term signal to noise ratio, is actually mis-used in this scenario and misleading. A better term to use is "signal to error" ratio. Thus your signal to error ratio increases when you increase bit resolution. If your sound card has noise at the -50dB level, the noise will still be at -50dB, no matter if it's 24bit or 16 bit that you record at. The only difference is that the noise will be more accurately recorded on a 24bit recording than a 16bit recording, thus will have less distortion added to that noise, due to the larger space between "Quantization levels" (that's a fancy word for the spacing between the numbers on the Y-axis).

So now I'm still wondering where the technical explanation is with the term "mashing down to 16 bits"?