What Mastering Chain are you Using?

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What Mastering Chain are you Using?

Postby har-bal » Sat Mar 18, 2006 12:26 pm

Hello folks

Everyone has their own mastering methods and tricks. How about sharing your ideas with others in this board.

Tell us what you are working with.


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Postby Mister » Sun Mar 19, 2006 9:06 pm


Before actually doing anything I'll listen thru once or twice just to hear what is there, of course...

My first step is to check imaging & L/R balance. Having that little detail taken care of helps a lot to get a nice solid soundstage. There's two ways of I check this: swapping channels; and hearing if the vocalist 'disappers' from the side channels (I'll explain that in more detail in another post if any one is interested).

Of course Har-Bal is next(!). I'll make adjustments based on my notes made during the listening session. Sometimes I'll try INTUIT first, other times after making my own settings, or INTUIT with adjustments - whatever gets me the sound I want.

Then it's into WaveLab. If necessary I'll apply a little noise reduction. My favourtie mastering chain generally is UAD's Precision EQ to roll off the extreme lo end (based on Har-Bal's analysis - and ears again!) and maybe some hi end sweetening if needed. This is followed by the UAD Fairchild with little or (almost) no compression, just some input gain. This is finished with UAD's Precision Limiter if I want a little character or Ozone's limiter if I want a little more 'transparency'.

If I don't need any hi end sweetening I might also insert the Pultec EQ just for the extra gain (~1.1 dB) and the 'magic' it makes just passing audio thru it.

Occasionally, I'll use the Fairchild to do mid-side compression to help fill out the soundstage.

Sometimes I'll use Ozone's mastering reverb for a bit of air - usually no more than 2 - 5%. Another approach to that end is to apply expansion (peak un-limiting) to open things up. (I master a lot of acoustic type material - folk, etc). Care needs to be taken if I also want some level gain because it defeats the purpose if the limiter is un-doing most of the expansion(!)

Finally, I use Ozone's M-Bit dithering most of the time with DC offsets removed.

One trick I use sometimes is to create a Mid and Side version of a track and process/analyze those separately in Har-Bal with a similarly treated reference track(s) to really nail a tough imaging problem.

I consider Har-Bal to be my most important tool right alongside WaveLab.

Hope this was helpful!
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Postby zumbido » Sat Apr 08, 2006 12:43 am

1 - Mix (PT on a Mac)

2 - Transfer to Har-Bal (on a PC)

3 - Tweak in HB

4 - Make an _eq file

5 - Open HB_eq file in T-RackS and use a bit of the compressor section with the multiband section.

6 - Open file from T-RackS in HB and take a look

7 - Transfer file to the Mac and open in PT.

8 - Edit beginning and end, put a fade-out occassionally.

9 - BTD with a dither plug-in set to 16-bit and convert file from 48kHz to 44.1kHz .aif file.

10 - Open up Roxio and drag files into playlist. Burn CD. Done.


I may repeat steps 1-3 many times and make 'corrections' to the mix based on changes made in HB - If HB tells me I have too much bass then I'll go back to the mix and work on modifying the low end.
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Postby HarBal » Sat Apr 08, 2006 8:01 pm

Hey Mister,

Your last trick envolving the Mid Side encoding is something I'm working on to make an integral part of Har-Bal. I've used a manual approach and Har-Bal like yourself to demonstrate to myself the worth of such a process. In my case it isn't so much an issue of fixing up imaging issues but resonance/masking issues. I've got stacks of commercial CD's that suffer from mid range resonances (mostly associated with vocals) that aren't apparent or fixable in the normal spectrum view. However, when you look at the side spectrum the peaks are obvious and easily corrected for. It has been quite a revelation hearing the difference that such a minor adjustment can make.


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Postby Bracelet Z » Fri Apr 14, 2006 1:34 pm

Hello, everyone!

Could You, guys, explain how to create a Mid and Side version of a track and process/analyze separately in Har-Bal? Coud You write this process step by step? I would really appreciate. I am also waiting for an updated version of Har-Bal with this function. Good luck, Paavo!

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Postby har-bal » Sat Apr 15, 2006 6:23 pm

Bracelet Z wrote:Hello, everyone!

Could You, guys, explain how to create a Mid and Side version of a track and process/analyze separately in Har-Bal? Coud You write this process step by step? I would really appreciate. I am also waiting for an updated version of Har-Bal with this function. Good luck, Paavo!

Bracelet Z

Bracelet Z

Instructions are below

Mid-side encoding is the same as is done for FM radio air chains.

Take L and R and mix together in equal proportions to give the Mid channel (equivalent to mono),

Take L and inverted R and mix together in equal proportions to give the Side channel. You can either do it as mono files or two stereo files. You can do it as two stereo files in the same manner as for the original LR thingy explanation. That is One stereo file (Mid) that has new L = L+R, new R = L+R and another stereo file (Side) that has New L = L - R, New R = R - L.

Mmix the two together you get back your original stereo track but with a 6 dB gain.

With these two stereo tracks harbalize them individually, the mid one in the usual manner and the side one with reference to the mid track and usually only minimalist adjustment concentrating on the elimination of bad resonances.

It can do wonders.

Earle and Paavo
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Postby HarBal » Mon Apr 17, 2006 7:05 am

I should add that the only processing you should be doing to the Side component with Har-Bal is controlling strong peaks with high Q notch filter cuts. You should aim for a very minimalist filtering. This will ensure minimal chances of audible side affects. If you happen to use wide Q filter changes you will drastically change the stereo image and you may end up with wierd effects like instruments moving with time.


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Postby tcatzere » Mon Apr 17, 2006 11:11 am


Where does the current Har-Bal "Air" fit into this M/S discussion -- or is that something all together different?

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Postby HarBal » Mon Apr 17, 2006 9:10 pm

It is related but not directly. HB Air will modify the relative amplitude of the side component in the stereo signal but it does nothing to it spectrally. This proposed M/S processing is specifically for controlling resonance issues buried in the Side signal that often don't show up in the summed average spectrum.

If the resonance is confined to the Side signal then you won't be able to correct it using normal means even though you'll hear it, especially in the far field. This change will allow you to easily fix another class of normally unfixable tracks. Applying HB air to tracks of this nature could possibly make them sound worse as it will amplify the side channel resonances, which explains why some tracks often take on a harshness if over "air-ed".


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my latest tool, www.brainworx-music.de

Postby wired » Thu Apr 20, 2006 12:40 am

this is cool, it helps solo and eq the mid and side seperately, i think with harbal one would also be able to fix the mixes given a mirror image of what to do, this device also has image shaping and monomaker (20-400hz) but i noticed its a little different in mixing, sometimes things could seem washed out if one increases the side mix too much, as its advized that kick, bass , ld vocal stay in the middle, but i like my bass and drums to go wider (these days with drumloops). in this device you can also output different levels for mid and side or mono sum, and stereo difference
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Postby MERLIN » Thu Sep 28, 2006 5:39 pm

Hi there,

Ive just recorded some vocals. My room is quite resonant although soon i will buy some sound absorbsion of some sort. I would like to try this mid sidechain thing on my vocal track, although im a little confused. I have the mono vocal track....Could you possibly explain it again, i coulsnt understad the instructions..thanks :)
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Postby lucky » Sat Nov 04, 2006 11:26 pm

Mix on my Tascam 2488. Run Mix thru Alesis Masterlink w/o tweaks and make a CD copy. CD in Computer to File. Then Har-Bal. Pretty basic at this point as I'm newbie. IntuitQ & a few tweaks til it sounds sweet. Pump up the Gain & add a little space. Then run back thru 2488 with Compression 0.0 to Masterlink then burn Redbook CD's.

Our first Har-Bal'd Song "There's Something About You" can be heard at www.soundclick.com/thelarsonbrothersband. Just click Music then click the song.

Check it out and critique if you will,


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Postby uncajesse » Sat May 26, 2007 12:57 pm

my mastering chain

First I will do a transfer from analog if needed. At this point I add 1sec silence to the beginning & end (or cut each track to that) and will run any noise reduction if needed. I use Adobe Audition v1.5 for all of my cuts & fades. Version 2.0 (at least the demo) seems to have much fancier graphics at the expense of speed. I prefer function over form. But v3 is out, I should try the demo one of these days.

Example of an original recording:
http://ictybtihky.com/harbal/examples/a ... iginal.mp3

Next I go into Har-Bal. My method is somewhat like the Emphatic technique from the help files, but instead of adjusting each individual 1/12th octave, what I do is make the center areas of the peaks & valleys (on the peak power plot line) be average across the overall curve that I'm going for. But I do not correct for any roughness between large and small deviations that may be right next to each other, unless it is audibly a problem. This helps to preserve the dynamics of improperly mixed sounds due to them being improperly harmonically balanced. This allows me to correct for that problem, without taking away from the original intent of the mixer in regard to that particular sound's harmonic relationships.

Here's an example of what I mean...
http://ictybtihky.com/harbal/examples/a ... arbal1.gif
I have smoothed out the averages across a few peaks, but I havn't smoothed out the difference between the peaks. I have allowed there to be extra large peaks & valleys, and also extra small peaks & valleys in the middle too. You can also see the frequency response of the filter, here:
http://ictybtihky.com/harbal/examples/a ... arbal2.gif

An example of a har-balized recording:
http://ictybtihky.com/harbal/examples/a ... harbal.mp3

Now I move into the finalization stages. My main goals in Har-Bal have been to smooth the balanced of the harmonics, but also to compensate for particular sounds/instruments/tracks that didn't have proper spectral/harmonic balancing which caused them to be improperly mixed into the recording. This is often done without priority for the overall larger audible spectral balance, because I'm using har-bal mainly as a tool to re-align those problems in the mix, but also the spectral balance from an engineering point of view as opposed to only audible. They both need to have compromises made while doing this. I don't want to create something that sounds so bad that it can't be compensated for in the finalization phase, or something that would create a sound that's too far from what the original artist had intended.

So for the finalization stage, I use (drumroll please) Winamp as my host. Before you think I'm crazy I think I should mention that I've tested the setup and found that it makes a 1:1 perfect copy of a WAV file when it's run through it using Nullsoft Disk Writer output. So quality is not an issue at all.

I use Nullsoft in_wave.dll v2.06 because the new ones only output 16bit or 32bit, not the input WAV file's bit-depth. It's the last version they released without the new "feature", and you'll need it if you plan on playing back 24bit WAV properly. You can get a copy I have here:
http://ictybtihky.com/harbal/apps/in_wa ... .v2.06.zip

I use the Adapt-X plugin chainer v3.61c
http://www.chronotron.com/content.php?p ... cts#adaptx
with "force 24bit floating-point" enabled.

I also use DirectiXer for making VST-only plugins available to DirectX hosts like Adapt-X.

Now for the plugins, I'll list the main ones I got to often, and I'll put an asterisk next to the ones that I only use optionally if needed. I'll also be linking to screenshots of my default starting settings, if applicable or not the plugin's own default. ;)

Voxengo PHA-979
this is a pretty simple plugin, it adjusts the phase of the channels, pan, and also has a gain control. it's useful for adjusting the center from analog transfers and/or acoustic recordings, and compensating for the slight phase tilt that usually happens with vinyl & from tapes recorded with badly aligned heads. a good technique is to use MSED to remove L+R (mid) and adjust the phase and pan until the bass/center is the most quiet.

Voxengo MSED
this is a pretty simple plugin as well, FREE and it can adjust the gain on mid & side.

my own custom 2-pass-band de-esser (*)
(sorry, no public information is available at this time)

Waves Renaissance Bass (*)
i rarely use this, but i use it on some tracks that just can't seem to get up enough gumption in the sub-bass department through EQing, without bringing up unwanted or otherwise "problematic" sub-bass issues. it's definitely resonant. :) I can't post a screenshot of my defaults because I created an XML preset and then edited it by hand, because there's about 4x more controls in most of the Waves preset files than what you see on the GUI. I basically took the default sound, and made it have more resonant bass yet to have much less mixed into the signal without having to reduce the "Intensity" knob at all.

Waves MaxxBass
i pretty much use at least a little of this on everything. It's really great, and accurate & honest to the bass input, when used so there's just enough to augment the bass, not replace it. I also don't use this very often in the rare cases that I use RBass. My default
http://ictybtihky.com/harbal/pluginshot ... xxbass.gif
(note: "HighPass" is set to 12db)

BBE Plugin v2 (*)
I use this sometimes near the end of tweaking all these plugins, if I can't seem to get the right snap or crispness out of a track with EQ. it's basically a distorted compander, with phase-sync on the harmonics. I very rarely use more than a setting of 1 on the "process" knob. I never use "lo contour" because it ruins the midrange.

Nomad Factory EQP-4
http://www.nomadfactory.com/products/an ... /eqp4.html
This EQ is amazing. It's a hybrid layout Pultec EQ emulation. I think if I had to pick any one plugin for an EQ, this would be the one. It sounds amazing even just running audio through it, much less actually EQing which is even more amazing. The lows are very warm and phat, yet it actually improves the solidity of the bass instead of smearing it. Both mids are anywhere from milky warm woodiness to silky warm musical articulation, depending on high high the frequency knob is set. And the treble is super duper smooth and graceful, very musical without over-accentuating individual tones. It can also be used for slight de-essing or to augment a de-esser. My default:
http://ictybtihky.com/harbal/pluginshot ... t_eqp4.gif

URS A-series EQ
Another amazing EQ. This is an emulation of an API (Neve) 550B console EQ. It really excels at bringing out, or pushing back, frequency ranges. But it's especially it great for bringing out high-frequencies. The Q of the bands increases as the gain goes up, so they start out very wide and musical. I almost always use band 3 & 4. And I sometimes use band 1 or 2 if the bass or mid-bass needs to be brought forward or backward. My default:
http://ictybtihky.com/harbal/pluginshot ... lt_api.gif

URS N-series EQ
Yes another amazing EQ. This is an emulation of a Neve 1084 console EQ. If i had to use one word for this EQ, it would be "musical!!!" The low band is the most amazing low shelving EQ I've ever heard. Even at 220Hz, it NEVER sounds muddy, no matter how much gain you give it. The midrange is equally as exciting, and it's soooo musical. If i had to define the coloring of the midrange with one word, I would call it "goodness". It really doesn't add anything to the midrange that isn't already there, it just brings out the good stuff. Generally the "center" frequency of this EQ is more towards the bottom of the actual filter, and it tapers upwards with plenty of beautiful harmonics. And the treble I usually don't use too often, but it is very good at bringing out tones in the upper treble, so I will sometimes use just a twinge of it, often at 12 or 16kHz, whatever is above the main "noteyness" of the treble, to add additional clarity if it's needed. The high-shelf can make some insanely crispy pitchfull treble, even with less than 1db gain, and it can reduce the same just as fast with -1db. So be very careful using the high-shelf, and make sure you have a DA & monitors that will do proper justice to it, or just forget about using the HF band on this plugin. My default:
http://ictybtihky.com/harbal/pluginshot ... t_neve.gif

Voxengo Elephant
This is a broadband brickwall peak & rms limiter. I have done extensive testing of a LOT of brickwall limiters, commercial and free, and Elephant surpasses them all by leaps and bounds, ESPECIALLY with regard to it's lack of any harmonic or inter-modulation distortions at all. Nothing else comes close to how transparent this is. Even Waves L2 looks disgustingly filthy by comparison. It is one of few digital brickwall limiters that can limit L & R separately, and perhaps the only with varying percentages of channel-linking. It also has Bob Katz's K-System metering, and a phase-compensated high-pass filter to eliminate sub-bass. It's best to remove un-needed sub-bass as early as possible in your chain though. My default:
http://ictybtihky.com/harbal/pluginshot ... ephant.gif

Apogee UV22 HR
This is considered to be the #1 bit-depth reduction algorithm. I would say that tied with it is POWR 2 which is available on the SADiE, and Daniel Weiss gear, among others... but i've always liked the sound of Apogee UV22 HR more, in every case I've compared the two. So I've always stuck with it, plus I can run it in real-time in my chain without having to route audio back over to the SADiE, keeping the convenience of using Winamp for my final stage. Settings I use are to dither to whatever depth is needed, usually 16int or 24int bit (remember Adapt-X is using 24bit 4 byte float). And I always use the Auto-Black function, which makes it only use as much high-frequency noise bias it needs to.

And then I render the WAV when I'm done tweaking. I also save each chain preset before rendering so I can re-render any version if i ever need to. Then I go into Adobe Audition v1.5 again. If it's 16bit, I'll reduce the bit-depth right away, using NO dithering because it's already been applied. I like to do my fades for 16bit while it's 16bit, because it doesn't add un-needed dithering noise that can sometimes be added to silence that's below the 16bit noise floor before dithering. I usually put 0.14 seconds of a special near-silence on the beginning and end, again using a plugin i made. The reason I don't use pure digital silence is because some crappy cd-players don't enable the DA until there's non-silence and then end up not enabling fast enough to catch the leading edge of the actual audio, which negates the whole point of putting a split-second of silence there in the first place.

And that's about it. For client proofing and final delivery I'll usually provide a 16bit WAV, two high-quality mp3s (320 & vbr), and sometimes 24bit WAV. I use Lame 3.97 with LameDrop front-end, using these settings:
-b 320 -q 0 --lowpass 22.05
-V 0 --vbr-new -q 0 --lowpass 22.05
But I'll provide whatever formats they request, of course.

and finally the fully mastered results on that recording...
http://ictybtihky.com/harbal/examples/a ... master.mp3
(slammed? well this actually only has about 2-3db of peak rms limiting, max. There is also a radio promo version with about 5db more headroom/punch and about 5db less "side")

I hope you enjoyed reading this, and I hope you can end up checking out some of the plugins I have mentioned. They all deserve attention, especially Aleksey Vaneev's Voxengo plugins. It's a one-man operation, and his plugins are top quality and very affordable.

later on,
Jesse Graffam

p.s. to zumbido, you should ideally be recording at 44.1 for CD, but if you need to support HD for archival & later re-purposing, I recommend recording at 192kHz if possible. but really, the quality of the output is mostly related to the quality of someone's DA filter, than the sample-rate. At least for PCM anyways, 1-bit streaming formats like SACD are not directly comparable in function.

but if you are forced to record in 48kHz, or need to record in 192kHz, you should re-sample to 44.1 BEFORE doing any of your mastering process. It may also be an improvement to use even a free audio editor on your windows machine for the final "cut & fade" step you are doing in pro-tools. Here's a decent one:

got rid of the SADiE, edited it out of this post too :) I also started using Voxengo LF-Max now & then right after the LF-Punch, and I'm demoing the Waves API EQs which DO sound great.
Last edited by uncajesse on Sun Jan 13, 2008 12:16 pm, edited 1 time in total.
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Postby HarBal » Sat May 26, 2007 9:54 pm

uncajesse wrote:my mastering chain

p.s. to zumbido, you should ideally be recording at 44.1 for CD, but if you need to support HD for archival & later re-purposing, I recommend recording at 192kHz if possible.

You should be wary of sample rate conversion between 48,96, or 192kHz to 44.1kHz. The ratio is not a simple one and as such, most sample rate convertors do a pretty bad job at it for such a conversion. If you want to record for CD then record using a sample rate that is an integer multiple of 44.1 such as 88.2kHz. By doing so you'll make the sample rate conversion much easier and much more faithful.

If you find this hard to believe I'd recommend taking a look at this web site, http://src.infinitewave.ca/. A perfect conversion would show a solid white line ramping up to half the sampling rate and no refected images above (aliases). There should also only be one line. Looking at the various options you'll see that many of the converters perform badly.

Certainly, if you want to produce a DVD and CD soundtrack use 96kHz but if CD is all you are working on then 88.2kHz or 176.4kHz is a better choice.


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Postby uncajesse » Sun May 27, 2007 1:31 am

Very good advice Paavo.
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