# Mixing/mastering process?



## Will Musser (Apr 20, 2007)

Hello,

I am currently not using any mastering software, I'm primarily composing with orchestral samples, along with some electronic stuff, spectrasoncis etc.

There is a clear difference when I listen to my tracks up against say a modern film score, or any commercial CD. I know samples vs. live instruments is a difference, but it seems like my tracks have a blanket drapped over the mix, muffled sound, even when using 24 bit samples.

My question is, what is the process from going from composing and recording a midi track/mix in Cubase sx to a pristine sounding mastered track? I know of WAves and Samplitude for mixering/mastering software.

I know its a general question but any insight for a newbie on mastering?

Thanks!

Will


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## sin(x) (Apr 23, 2007)

dach @ 2007-04-23 said:


> If you start slamming a poorly balanced mix with a limiter you will end up with a disaster...



Seriously, this should be framed!

If you have to resort to multiband wizardry or excessive limiting to keep your mix from sounding weak and dull, *something is very wrong with your mix*, and chances are you'll be better off ditching the stuff from the main buss and going back to the mixing stage if you have the time. Of course loudness is desirable, but if you have to slam your mix against a battery of dynamic processors to achieve it, this is a severe warning sign that you're about to sacrifice all your dynamic range and punch for it, which in turn leads to the infamous harsh, ear-fatiguing hypercompression syndrome. Once you have a good mix with a working frequency distribution and well-balanced dynamics, you can very often go with just a tad of peak limiting to get a loud AND punchy signal.


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## lux (Apr 23, 2007)

I agree, limiting is just the end of a process.

Here are a couple of philosophic guidelines i follow when i'm not just too lazy (thats kinda rare)

- Frequencies: dealing with samples and synths you'll find that every sound you put in the mix tends to get an unnecessarely range of frequencies. Most of timbres have a natural range of freqs, and if you get rid of unnecessary freqs you'll leave a lot of space for other instruments. 

- When dealing with orchestral samples i always find an horrible fog at about 110 hz, expecially when using contrabasses and cellos. Trying with an eq or compressor to balance this area helps a bit in my experience.

- Balancement: if you listen carefully at low volume you'll probably see that you left a couple instruments too high. If you want to slam your track you have to balance those otherwise no limiting will apply without distortions and craps

-Most mentioned muddyness area is at about 300hz

-cymbals are usually the worst enemies of my mixes slamming

- As Midphase pointed out using different listening devices helps to understand whats up with your mix

Well, nothing really useful, but thats what i usually try to do to have a little bit of a decent mix.

Luca


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## hv (Apr 23, 2007)

If money were no object, this would be my dream piece of mastering gear:

http://www.legendaryaudio.com/overview.html

Till I strike it rich, I've been getting by with trying to do as well as I can in the mix. And sticking to point source mic techniques (mono or x-y/m-s stereo) and source-based reverbs (vss, ray space, and convolution) to minimize imaging and phasing issues.

Howard


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## Will Musser (Apr 23, 2007)

Wow, thanks everyone for the info, very helpful.


midphase, you raise another point, the bass is killing me. I'm using some m-audio SP B8 or something,I forget what the name of them is, they are about 4 years old, but my tracks clip at a very low volume level, and I'm seeing that its the bass thats the reason. Thorugh my moniters I can't seem to get the bass loud enough, but then I go play it on my entertainment system, and the bass is too loud.

So, any reccomendations for moniters?

Thanks!


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## Will Musser (Apr 23, 2007)

Here is a recent track, I've done that is having the clipping issue, volume too low, yet clipping.

http://www.willmusser.com/chasingtherhino.wav

Any ideas?


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## dach (Apr 23, 2007)

hv @ Mon Apr 23 said:


> If money were no object, this would be my dream piece of mastering gear:
> 
> http://www.legendaryaudio.com/overview.html



My dream piece of mastering gear would be a shapely hot blonde with a good room, monitors, plenty of outboard and a Sadie system who was on call 24 hrs/day and worked for sex.

If she actually knows anything about audio, that's a definite plus....


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## sin(x) (Apr 23, 2007)

Will Musser @ 2007-04-23 said:


> my tracks clip at a very low volume level, and I'm seeing that its the bass thats the reason.



The problem might be in a range even lower than the bass you're hearing - if any of your signals has significant energy below 30-40kHz (microphones picking up rumble are a common cause for this), it can greatly decrease your headroom, thus making your mix clip very early, without you ever noticing what's wrong (especially when listening on monitors that don't have a chance to reach that far down). It's often good practice to insert a steep highpass filter at 20 or 30 Hz into your main buss to get rid of useless sub-bass frequencies that clog up your headroom (of course, your mileage may vary when you're mixing trance/techno or other styles that rely on sub-bass).


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## hv (Apr 23, 2007)

Will:

Gave your mix a listen and look and it strikes me that the problem is simply the levels being too high. You're clipping. All your clips seem to be on bass drum hits, but if their levels are what you want relative to the other instruments, you have to just lower levels on the whole mix. Assuming these are sampled drums and don't have a rumble problem. You might try lowering the entire mix by say, 6 db, then popping it into a wave editor that lets you read peak stats to see how much headroom there is. Then bump the mix back up just enough to leave .1 db or so of head room. I always check my mixes in a wave editor to make sure I don't have inadvertent clips. It also pays to eyeball track meters during playback to make sure there are no track-level clips which are way harder to locate in the final mix.

Another approach is to note the exact times of the clips/peaks and go into your instrument tracks and try and duck other instruments down at the point you want those drum hits to dominate. If that's not enough, try putting a velocity/automation envelope on the drum track and draw in a little dip on the errant peaks to tame them. I usually do this in the audio domain after freezing the instruments. This is kind of a manual compression technique that is useful when you only have a limited number of dominating peaks to deal with. I think it's the best approach, as opposed to using compression plug-ins, because it avoids stepping on other parts of your mix that don't need it.

Howard


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## José Herring (Apr 23, 2007)

This is the exact same problem I use to have. I can tell exactly what's going on. Seems that you're trying to push the levels in order to feel the impact of the music. What's probably the cause is, as Midphase suggest, the monitoring levels are not set correctly so you have to push up the fader levels in order to feel the impact of your music.

It's a pretty easy thing to do to set the monitoring levels right. Somebody on this board once walked me through it. If this is the problem then I can point you to that thread.

Mixing at the right level I think is crucial. Then turn down the audio levels and then master to get the most signal onto the "tape" as they use to say. But if you mix too hot due to monitoring too low then you'll experience all the problems that you have here. Setting the monitors at the right db level will fix this tendency to push the faders to the max for impact.

best,

Jose


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## Lpp (Apr 23, 2007)

Ok, I gave the piece a listen on my lame pc-monitors, as this most often reveals all the little problems and I must say...

neither is this thingie not loud enough, nor it is too muddy. I find it quite right. 
The only thing, that makes for some mud is one low drum, that I can´t identify. I first thought, it could be the timpani, because it has a similar frequency. It always comes at the loudest passages. 
My suggestion would be to either eq this sucker or make it quieter. Then everything should be fine.

Hm... you sure, I didn´t hear a newer version of this piece ? For me it sounds not anything like you described ~o)


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## Synesthesia (Apr 23, 2007)

As well as the great tips mentioned so far (especially tuning your mix room as best you can) there is the issue of 'leve to tape' in DAW land.

I believe I got this from Bob Katz' book, which I recommend, but basically you need to leave plenty of headroom in the DAW on your individual tracks to get a decent mix out of the mixbuss.

Its the exact opposite of tape in that respect, and you aso reduce the 'retranslated overages' in transients as well. (ie, your reconstructed analog waveform exceeds 0db digital full scale as a result of the 'calculation error'.)

Leave lots of headroom and use plugs like DAD Tape and Valve to get a nice saturated sound from a channel.

But, again, get the Katz book, its chock full of amazing insight and experience, I keep it in the lavvy as thats the perfect place to meditate upon its wisdom. :mrgreen: 

Well worth the 20 quid or whatever it cost.

Good luck,

Paul


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## sin(x) (Apr 23, 2007)

josejherring @ 2007-04-23 said:


> Setting the monitors at the right db level will fix this tendency to push the faders to the max for impact.



You know Bob Katz' K-System? It's a calibration and metering system developed for exactly this purpose. It consists of 3 scales - K-12, K-14, K-20 - each of which matches an internal dBFS level to a physical sound pressure level (83 dBC at -12, -14 and -20 dBFS, respectively). K-14 is meant for typical pop productions, K-20 for classical, jazz and film postpro. You pick a scale for your job, switch your master VU meter to it (so that 0dB is at the reference point), and adjust your monitor gain to match.

Ever since I calibrated my monitors to these scales, I'm experiencing that most gain-staging actions I did with the help of VU meters before now come kind of naturally. 83 + 14 = 97dBC is _really_ loud, so I'm thinking twice before I push anything into the red - which in turn results in rather dynamic mixes with lots of headroom, and a natural barrier against the temptations of hypercompression. Nifty!


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## José Herring (Apr 23, 2007)

Haven't read that book. I should get it. Basically with the help of a forum member, David Robinson, I set up my monitors to mix at 86db on my handy Radio Shack sound level meter. I never let the Daw master bus get above -10 or 12 for softer pieces -8 for medium and -4 for crackling loud pieces. In my daw mixer my peaks never see red and my soft instruments never go below a certain point ( i think it's -30db or so. Not sure I just set it up in my Daw to turn a certain color when I get too low.) 

What has happened as a result is that all of a sudden your mixes themselves turn into a dynamic expression.

I'll certainly pick up Bob Katz's book for more insight.

Jose


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## Niah (Apr 23, 2007)

I think it's important to state what type of music are we talking about here.

In my experience orchestral and film music don't require much compressing and hard limiting. The use of these plugs should be kept at minimun and when overused kill the dynamics of the piece. 
If you want your track to be loud you should mix it loud to begin with and then later use a HL just to make sure some frequencies don't go over 0db. However, if we are talking about orchestral film music I don't think it should as loud as pop stuff.
As for the muddyiness/muffled sound it's a question of EQing.

I too recomend oZone3.


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## midphase (Apr 23, 2007)

> Thorugh my moniters I can't seem to get the bass loud enough, but then I go play it on my entertainment system, and the bass is too loud.
> 
> So, any reccomendations for moniters?




Still everyone totally ignores the room.....THE ROOM IS HUGELY IMPORTANT PEOPLE!

Sounds to me like you don't need new monitors....you need to do some acoustic treatment and tuning in your listening environment. 

Misconception #1 - Tuning your room and keeping sound in/out (aka acoustic insulation) are the same thing.

They are not! Just putting foam or egg crates on your walls does not necessarily make the room accurate. Acoustic tuning means utilizing sound dispersion surfaces to avoid reflections, using bass traps, ceiling mounted diffusors, carefully placed acoustic foam is key areas. Usually the best way to do this is to have your room professionally analyzed for a (relatively) flat frequency response in and around your primary listening position. Since this is very expensive for most people, the next best thing is to do research, and slowly improve your mix environment by mixing, doing acoustic treatment, mixing some more....listen for the differences, learn your room and so on.....this process might take months or even years until you feel comfortable enough with mixing in your room where you know how things will sound everywhere else.


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## Moonchilde (Apr 23, 2007)

You can get equipment that will EQ your monitors to match your room acoustics so you get a "true" and "flat" sound. I bought a receiver that does this, I'm cheap and I use my monitors for everything so I have them running out my receiver instead of my audio card. But the nice thing is, everything just sounds a lot better with the auto EQ based on your room's acoustic profile. I chose this because it is really impossible for me to do room acoustic treatment by remodeling an apartment that doesn't belong to me. Oh, and it was a lot cheaper too.


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## midphase (Apr 23, 2007)

I think that if you do that, you need to be very conscious of where your listening position is and make sure that it is the same each time you mix since the room is EQed based on where the measuring mics are placed.

Sometime it's a matter of inches and not feet....move just slightly away from where the measurement was taken and your accurate room might disappear.


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## Leo Badinella (May 8, 2007)

Will Musser @ Mon Apr 23 said:


> Here is a recent track, I've done that is having the clipping issue, volume too low, yet clipping.
> 
> http://www.willmusser.com/chasingtherhino.wav
> 
> Any ideas?



Ok, I'm kinda late to the party but here's what I think:

I found clips at almost every big hit on the track, if you can't fix this in the mix what I would suggest you do is use a soft clipper. There's a free one for PC called GClip. And I know there's an embedded soft clipping algorhythm in PSP MixTreble, so you might wanna experiment with that. The only downside with MixTreble is that you can't control the knee (amount of softness if you will, or when the plug starts working level-wise).

Within the mix I would suggest you use some kind of saturaton plug or limiter, maybe a tape emulator, but just on the offending instruments.

...I would show you how this sounds but I can't attach mp3s so I guess thòÌG   XOÌG   XOžÌG   XOŸÌG


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## SvK (May 21, 2007)

If you are doing pop / rock / radio music....then mastering plugs are great......since you can get your music LOUD and give it that "compressed", "tite" sound...... 

however, at the expense of dynamics......and dynamics are what orchestrals are all about..... 

Now in film music it is not necessary to turn your piece into a "brick" since it will rarely be mixed that loud beneath the dialogue, sfx anyway...... 

Mush better to focus on the eq's placed on your busses for the violins, violas, basses etc, etc...... 

ps: On a further note many, many of the leading producers and engineers out there are trying to get the record companies to stop "L1 ing" there music to kingdom come......it causes ear fatigue, and robs the music of all it's natural dynamics... 

As an example try this: 

Bring an aggressive new "Green Day" song into your DAW...Now bring "Back In Black....Ac/DC" into your DAW as well.........Now when you play them back to back "Green Day" song will sound 8db louder then "Back In Black" ....now lower the "Green Day" song to be the same volume as "Back In Black"...... 

Now turn up the volume of the speakers and play songs back to back 

The 80s mix of "Back In Black" sounds WAY bigger.... 


SvK


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## SvK (May 21, 2007)

To put this into perspective........ 

It all really started in the early 90's.....Tools like the "L1" from waves came along......and record companies realised they could make their releases sound "louder" than their competition's on the radio..... 

Because if their song starts playing and it sounds "louder" then the song before it, the listener will find it more noticable (exciting, powerful)........but it's an illusion.....it's being "squashed" in order to be louder, so the percusion becomes less "punchy", less "percussive"..........this is bad, very bad. So now that this song is sounding louder, the competition squashes their song even MORE!!!! And it's been going that way since the early 90's.......Now most mixes just sound like lifeless pancakes.....

I hope this makes sense. 

SvK


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## SvK (May 21, 2007)

In an ideal situation......

If ALL the record-companies, backed of from the "L1, C4, L2, L3" by 6db or so....and we the listeners just turned up the volume on our stereos......music would sound way, way BIGGER, better again.

what a crying shame....

SvK


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## ComposerDude (May 22, 2007)

Good points, Steve. Overcompression is neither good art nor good science.

Here's what some gurus of radio station broadcast processing have to say about the myths implicit in the "loudness wars"...and why if you don't squash your mix, it will sound MUCH better through the broadcast audio chain...

http://www.orban.com/support/orban/techtopics/Appdx_Radio_Ready_The_Truth_1.3.pdf (http://www.orban.com/support/orban/tech ... th_1.3.pdf)

-Peter


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## SvK (May 22, 2007)

ComposerDude,

great write-up...thanx for posting this truly revealing link.............


ps: There is a similar forum on "Vienna"....I'm posting this link there 2.

SvK


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## Ned Bouhalassa (May 22, 2007)

Is there a difference between the processes (comp, eq) used on music by radio stations and those used by television broadcasters?


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## SvK (May 22, 2007)

Ned,

One thing is certain....There is nothing more over-compressed and Louder than a movie trailer 

ps: (I do think over-compression has it's place and uses....short bursts of it can make for a great effect.....Squashing the sh#t out of room-mics for drums and adding that to the uncompressed close mics is another example)........just not on masters / entire mixes / albums... We are destroying music.

SvK


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## Synesthesia (May 22, 2007)

Ned Bouhalassa @ Tue May 22 said:


> Is there a difference between the processes (comp, eq) used on music by radio stations and those used by television broadcasters?



Does anyone know? I'd love to know.. I'm going to see if I can find out from a few of my contacts in TV. If I find anything out I'll post it.

Peter - thank for that link - very informative. I did some radio work years ago for the independent radio chart show here in the UK, the guy who helped me mix it was the guy who set up Capital Radio's compressors: the mix sounded pretty wierd in the studio but when I heard it on the air it was a revelation! He really knew how to trigger the right 'moves'.

Kays - do you know of any little app that you can run to test tone your room with a mic? It would be great to find an audio 'test tone' file that has a full run of tones - I guess if you record it back into your DAW you will be able to see the peaks and any problem frequencies.. 

I have to say I think my room sounds pretty good (for its shape) but I have placed my traps pretty randomly (taking out as many corners as possible for the bass trapping and a few panels parallel to the walls to capture some HF energy, with a litte inspiration from the realtraps site - a more scientific approach might tighten it up a bit more..

Thanks,

Paul


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## Synesthesia (May 22, 2007)

I just realised I could spend 15 minutes making a test file with the protools signal generator and a bit of automation.. 

If anyones interested to grab it I'll whack it on a free file server (any suggestions?) 

I'll run it tomorrow and record it back through a U87 and see what the results are and report back if it gives any useful info!

cheers

Paul o-[][]-o


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## Ned Bouhalassa (May 22, 2007)

Congratulations, Kays! o-[][]-o


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## Synesthesia (May 23, 2007)

Hey Hannes..

http://rapidshare.com/files/32890745/testtones.aif

It was quite revealing - I have two problem LF areas, quite small, one dip and one boost, around 130-140Hz..

I put the mic in the sweet spot and ran the output from the speakers back into protools.

I'm going to run a pink noise test also as suggested by Kays, but only from the listening position - theres going to be areas of cancellation in other parts of the room but I'm not bothered about that, I want to tune the sweet spot as best I can..

Any thoughts on the relevance of the polar pattern of the mic you use?

Cheers,

Paul


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## Ned Bouhalassa (May 23, 2007)

One thing to have handy in this case is your microphone's frequency response chart. I just did the test with my Rode NT1000, and the returning pink noise looked great in Logic's Multimeter until I noticed that the mic has a boost in certain HF areas, which means that my sweet spot has some holes in those very same areas... :roll:


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## ComposerDude (May 23, 2007)

The measurement-style mics I've seen are omnidirectional mics with a capsule at the very tip of a somewhat long cylinder. Cardioid and other restricted patterns have a polar pickup pattern varying with frequency. So omni would be your first choice if you have such a mic in your collection, and more specifically: small single-diaphragm omni.

Do your pink noise testing at LOW volumes unless you want to risk your woofers. It contains a surprising amount of low frequency energy on a continuous basis that doesn't let the woofer coils cool. I learned this the hard way: it's the only time I ever burned out woofers. And since the objective of pink noise is to calibrate room response WITHOUT exciting room modes, low-volume testing (as long as it's reasonably above the noise floor) works fine.

-Peter


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## hv (May 23, 2007)

If you've ever worked with Voxengo stuff, their Deconvolver plugin has a sweep generator built in and the demo version allows 3 sweeps per session. But that opens the door to a totally different way to treat your room. Convolution. If you could take an IR of a studio you love, you could then cause yours to sound just like it, monitors and all.

Btw, DBX makes and inexpensive $100 RTA omni and the similar Behringer ECM8000 goes for around $50.

Howard


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