Guys, the keys to a good mix? Can anyone explain frequencies to me?

EQ

One thing that is *ESSENTIAL* to a professional sound is mastering the parametric equalizer. Graphic equalizers just don't cut it for recording; in fact, I *NEVER* use one personally.

Parametric EQ is easier to use than you think, but it can appear tricky to master at first. It's not--just use your ears and don't over EQ.

KEEP IN MIND THIS GUIDE MAKES MORE SENSE IF YOU OPEN UP A PARAMETRIC EQ AND MESS WITH IT A LITTLE BIT WHILE YOU READ THIS!

This is a quick and dirty guide to parametric EQ.

First off, parametric EQ's have multiple bands--typically 4 or 6 bands. Each band is independent and can have its own individual settings. Most parametric EQ's have a number of MODES or FILTER TYPES available to them for each band:

HIGH PASS FILTER: will not affect freqs *higher* than the center frequency--in other words this cuts out lower frequencies (the highs PASS thru--get it?).

LOW PASS FILTER: reverse of the high pass--the freqs *lower* than the center frequency are unaffected--this cuts off high frequencies.

Both high pass and low pass filters have something called a *roll off* which may or may not be user definable; a roll off will determine the slope of how the frequencies are reduced--such as 6db per octave, 12db per octave and so forth. The greater the db reduction the more frequencies are reduced.

High and low pass filters are usually only available on the ends of the parametric EQ bands. Thus, a 4 band parametric could have a high pass filter, 2 band filters, and a low pass filter as its options.

NOTE: both low and high pass filters *ALWAYS* are used to cut frequencies--these cannot be used to boost.

SHELF FILTER (LOW OR HIGH): affects ALL frequencies from the center frequency and upwards (for high shelf filter) or below (for low shelf filter). Use carefully and sparingly. This is basically a relative of the high/low pass filters but contains no roll off.

BAND FILTER: this is the "typical" mode you will use--this will accent or cut a certain range defined by the user (see below).

After the mode, which 90% of the time you will be using a band filter type of mode, the next important thing to look at are the actual controls of the parametric eq--the center frequency, Q, and gain.

CENTER FREQUENCY: this is the epicenter of the where you are applying EQ at. Usually ranging from 20 hertz to 20,000 hertz (20 khz). This is just the center of your eq adjustment, other frequencies will be affected.

Q: this determines the width of the eq around the center frequency. The higher this number is the narrower the range. Very narrow boosts can sound "ringy" and actually go into a (bad sounding) self-oscillation due to the feedback used to create the boost.

GAIN: the "height" or "depth" of the equalization. The gain, which can be positive or negative, determines how much cut or boost you are using in that frequency range.

In general, keep all gain cuts/boosts within 6db. Most of my cuts are under 2 or 3 db's these days. If you record sounds to be *exactly* what you want you don't have to mess with them very much--resulting in a much cleaner, pro sound.
 
PHASE

Right. There are two different things going on here: phase and polarity are only the same thing at one specific special-case point, really. Phase is a much more complex topic.

When two signals have exactly opposite voltages at each and every point in time, they are said to be "opposite polarity". They also happen to be 180deg out of phase with each other: that is the most gross phase error possible, and it happens to apply equally at all frequencies for this one special case. Add 'em together, you get zero: everybody knows that.

However, when you introduce *time* delays by spacing mics apart, the simple "opposite polarity" special case doesn't occur: because the sound takes time to go from here to there, and the wavelengths differ according to frequency, you have differing amounts of phase shift at different frequencies. This "different-phase-shift-per-frequency" thing is what gives you the "phasey" sound with a pair of mics at different distances from a source: you are creating a comb filter, where some frequencies are 180deg out of phase and cancel, and other frequencies aren't quite out and don't quite cancel, and others still are exactly in phase and *add* instead of cancel. What you really work with in mic placement is this parasitic comb filtering: it comes for free, because of the time delay.

Let's work an example, just for shits and grins. Sound moves about 1100 feet per second (close enough for this example). So let's stick a pair of mics on a guitar amp: one right up against the cone, and one back a few feet. The picker graunches a chord, and the resulting signal has all sorts of components at different frequencies, from the lowest string's fundamental at 100Hz, say, all the way up to high harmonics up at 10kHz or so.

The wavelength of a 100Hz sine wave is 1100/100: 11 feet (nice, round number, eh?). So theoretically, to perfectly cancel that 100Hz out between the two mics, you'd put the second mic back 5.5 feet: one half wavelength. Ignoring reflections and all that unpleasant real-world dreck, when the pressure was just hitting its max positive value at the close-in mic, it'd just be hitting its max _negative_ value at the mic 5.5 feet away: one half wavelength, 180deg phase shift. With me so far?

Now let's look at some 1kHz stuff in the same signal. That 1kHz stuff has a much shorter wavelength: 1100/1000, or 1.1 feet (start to see the relationship?). So to completely cancel that 1kHz, you'd only have to move back .55 foot: 6.6 inches. The perfect cancellation distance for any given frequency is _not_ the perfect concellation distance for any other frequency, because the wavelengths differ. Make sense?

If we move the second mic back that 6.6 inches behind the first, we have 180deg phase shift (cancellation) at 1kHz, and only 18deg phase shift at 100Hz: not much at all, and very little cancellation. The 1kHz stuff covers 10 wavelengths in 1 single wavelength of 100Hz. That's the key!

So phase shift is a function of frequency, when you're talking about spacing between mics. In the first example there, that perfect cancellation of 100hz at 5.5 feet would give you a perfect _doubling_ at 200hz (360deg phase shift, or perfectly in phase), and a perfect cancellation at 300hz (540deg phase shift, or effectively 180deg out of phase) and so on: so you end up with a frequency response that has a series of peaks and notches in it. Thus the term "comb filter".

Changing the spacing moves the notches around, and that's what you are doing when you play with mic placement. It also changes the contributions of reflections and all sort of that real-world stuff sweetnubs mentioned that is really a damned sight more important than this nerdy theoretical shit. So you never get the precise cancellation or reinforcement that this contrived and oversimplified example might seem to predict: the comb filtering can be pretty subtle. But that's how it works.
 
This has been posted many times in several different places. Seeing as I will be bringing these steps up from time to time, and there is a constant influx of new people, I will post it here as a referrence.

1. Mixing is an attitude.
2. If the song sucks, the mix is irrelevant.
3. Working the room, keeping people happy and relaxed is half of mixing successfully.
4. Putting everything proportional in a mix is going to make a shitty mix.
5. Gear are tools in a mix that make life either easier or more difficult, they are not what makes a mix good or bad.
6. A mix can be great and not have great sound.
7. If nothing about the mix annoys someone in the room, the mix is often times not done.
8. Mixing can not be taught, it can only be learned.
9. The overall vibe of the track is much more important than any individual element.
10. Just because it was recorded doesn't mean it needs to be in the mix.
11. Be aggressive.

What can I say? My steps are kind of like a Marshall amp. They go to 11.

Mixerman
----------------------------------------------------------------------------------And I once posted this:

The Dreaded Mixdown

This is where many new recordists fall down. It's one of the hardest things to get right, but there are a few things you can do to help get your mixes closer to where they should be, right from the start. (MixerMan, who gets paid big bucks to do this, will hopefully jump into this thread at some point.) It requires a different mindset from tracking and arranging. It also requires that you not be married or in love with any one part in the song.

Tip 1. Get as far away from the song as possible before you try mixing it. Don't try to do a mix right after a tracking session. Your ears are fried, and you're too close to the song right now. Objectivity is the word to remember. Wait a few days or even a week or more, if you have that luxury. Yes, some people can do a good mix right away, but that usually takes years to acquire that skill. If you haven't been doing mixes for many years, you ain't one of those people, so wait.

Tip 2. Mix low. Yes, cranking it sounds cool, but it will also introduce more room reflections and give you a warped picture of the sound. Crank it when you think you've got the mix nailed, but keep it low for as long as possible.

Tip 3. Listen to the song, not the tracks. The biggest mistake new mixers make is soloing each track and making it sound full and rich by itself, then they wonder why the whole thing sounds bloated and muddy. There are several methods that work to construct a good mix. You can start by bringing all the faders up, with the pan pots centered, and all effects turned off, or you can decide what the key element in the song is (the vocal, for example), and start working from that. Different engineers use different methods.

Tip 4. Build a box - a small stage in your mind. Imagine a stage. You control where the player appears on that stage. Panning lets you control left to right placement, volume and reverb lets you control front to back, and eq lets you control the frequency blend (low to high).

Tip 5. Resolving conflicts in the mix is the single biggest problem facing a mixer. You'll always find several tracks competing for attention in the same frequency range. The kick competes with the bass. The bass competes with the low guitars. The guitars may be competing with the vocals. The keyboards are all over the place. It becomes an even bigger problem for most people when they solo a track and work to make that instrument sound as big as possible. Bad move. All the instruments hafta work together and a particular instrument has to sound good with ALL the other instruments.

For the good of the song, some of the bottom end on the bass or the guitars may have to be eliminated. Yes, the instrument may not sound good when it's soloed, but it will blend in better when you listen to all the tracks. It's up to you to decide which instruments need to be shaved, but if you concentrate on the song first, it will start to become more and more obvious what needs fixing.

Tip 6. Take frequent breaks and get away from the music for a few minutes. Rest your ears. If you're doing it right, it's the most demanding part of the whole recording process. You are literally listening to ALL the instruments at the same time, following them all at once, and it's easy to burn out. Wanna see an engineer really blow up? Try talking to other people in the control room while he's trying to work on a final mix.

There's a lot more, but we'll save it for another day, or wait to let others weigh in on this most difficult of all subjects.

Harvey Gerst

----------------------------------------------------------------------------------
And Mixerman added:

Ahhhh.. my favorite subject. I could speak for hours and hours on mixing. Harvey's tips are great. Defenitely valuable to the beginning mixer.

What can I add? Well let's start with the fundamentals of what you're working with. It's allot to digest, particularly with Harvey's list, and it should probably have it's own header, but I'll put it here anyway.

Barring 5.1, you only have 2 speakers to work with. But we live in a 3 dimensional world. So we're basically creating an illusion so that a mix sounds 3 dimensional. Let's call this a spatial illusion

When mixing there are 5 planes of spatial illusion. Level, panning, frequency, spatial perception, and contrast. These five planes are all used to create space in a mix.

Front to back: (Level)
Level gives an element of a mix it's own space. Compression on individual channels helps keep the level so that it doesn't disappear in the mix. A loud instrument will appear forward, or towards the front. A quiet instrument will appear to be back or further away.

Left to right: (Panning)
Panning allows you to give an element of the mix it's own space. For instance putting a guitar part hard right keeps it from washing out the vocal.

Up and down: (Frequency)
Frequency is the use of EQ to boost or cut frequencies that either muddy or clear the mix up. For instance 250Hz-700Hz are fairly muddy frequencies, and if you have too many instruments using this frequency range the mix could be muddy. Everything in an arrangement or mix should have it's own unique fundamental frequency space.

Far and near: (Spatial Perception)
Spatial perception is the use of reverbs, chambers, plates, delays, far mic placement, etc.. to create the illusion of space in the mix. An instrument with allot of reverb can sound like it is placed in a large hall. An instrument or a vocal with a long delay, can sound like it's in the alps. An instrument that's completely dry, will sound like it's in a small carpeted room, right next to you.

Sparse to dense: (Contrast)
Arrangement is the use of muting, and altering the recorded arrangement to create space where it is needed to accent the more dense parts. The use of density to contrast sparse is great for creating the illusion of dynamics in a mix, within minimal dynamic range. The use of a limited dynamic range makes for better listening in more casual environments, where there tends to be external noise.

All 5 of these planes work together to create the illusion of space in a mix. One is no more important than any other in general, although one or two of the planes could prove to be more useful in a given mix. Not all are a requirement for a great mix either. For example, your mix should to be able to break down to mono, and still be a greqat mix.

Mixerman
 
Your path bays come "half-normalled" which is perfect for several applications. Unnormalled will be used for some applications. Fully normalled will not be used at all. So here is the deal.

The purpose of the path bay is two. One, the make a normal signal route for you. Two, to give you the ability to access that signal path to have the option of going somewhere else with the signal with or without returning it to it's destination. Sound confussing? It might but I will explain.

In a "normalled" configuration, the top and bottom jacks are connected together. This would be used to say go from the output of your console to the input of the tape deck. You don't need to do anything to complete this connection except to hook the cables up to the back of the patch bay. The proper way of doing this is the plug the output of the console to the top jack, and the input of the tape deck to the bottom jack.

Now, you can use a patch bay to run from the output of the tape deck to the input of the console. You would do this by connecting the ouput of the tape deck to the top jack and the input of the console to the bottom jack.

Now that you are all connected, you can start using the patch bay to insert something in between the destinations.

Let's say that you are tracking and you feel that you need to hook up a peak limiter before you get to tape. All you would have to do is on the front of the patch bay that is configured for console to tape deck, take the top jack for whatever channel you need the limiter on and run a cable from that channels top jack on the patch bay to the input of the limiter. From the output of the limiter, you would heek up a cable to the bottom jack of that channel on the patch bay. Vola!!! You have now inserted a limiter into the signal path.

You may notice something though. The limiter will not be inserted untill it's output is connected to the bottom jack on the patch bay. That is because the "half normalled" connection at the patch bay that connects your console output to the input of the tape deck is not broken untill something is plugged into the bottom jack on the patch bay.

So, let's say that not only would you want channel one on the console to go to track one (you already have this hooked up on the patch bay by following the above wiring) but you also want to send that track to something else like a drum module or something. Well, you can access that tracks source from the console via the top jack on the patch bay without affecting that source from getting to the tape deck. You see, the top jack on from just gives you access to the source signal. The bottom jack is an interrupt to the destination. using them both acts like an insert. Cool!!!

Next. Un normalled.

Usually you use unnormalled configuration for signal processors such as compressors, gates, limiters, etc......Unnormalled means that the connection is not made between the top and bottom jack on the patch bay. The reason for this is because processors don't like to have to output feeding back to the input. So, you have to have a way of accessing the input and output of the processor without the two being halfnormalled.

Usually (and this is the case with all the above named patch bays in this thread) you unnormal the patch bay by simply taking the curcuit board and turning it upside down. It is that simple. Now the output of a device will not feed the input. Actually, by doing this the input feeds the output which is okay.

So let's say you have a Re an 48 point patch bay, one 8 track machine, two stereo processors, and one stereo effects processors. Here is what you would do to make it all fly on the same patch bay.

Tape output's 1-8 of the console would go to the top jack (on the back of course) 1-8 on the patch bay. Input's 1-8 on the tape machine would go to the bottom jack(you got it, on the back) or the patch bay. Bam, 1-8 is all hooked up. The outputs of the console will go to the input of the tape deck automatically without any patching.

Next. Output's 1-8 on the tape machine would go to the top jack 9-16 on the back of the patch bay. Tape return 1-8 on the console would hook up to the bottom jack 9-16 on the back of the patch bay. Bam!!! Now your tape deck will automatically go to the console without any patching.

Next. 17-20 on the patch bay need to have their curcuit boards turned upside down. Now, you would use 17 and 18 for channels one and two of the first processor, and 19 and 20 for the second processor. Remember, processor input to the top jack, processor output to the bottom jack. Bam, processors are working.

Next. Take your aux sends one and two from the console and go to the top jack 21 and 22 of the patch bay and the effects unit input will hook up to the bottom jack of 21 and 22. Bam, automatic effects sends from the console to the effects unit without patching. Next, effects unit left and right out to top jack 23 and 24 on the patch bay, and whatever you would normally use for a effects return on your console to bottom jack 23 and 24. Bam, instant effects return.

So, if you wanted to compress track one before going to tape, plug in a cable from the top jack of one and run it to the top jack of 17 (left channel of your first processor) and hook up another cable from bottom jack 17 to bottom jack 1. Bam!!! Your compressor is "inline".

Now let's say that you are mixing instead and you want to compress track one.

You would hook up a cable from top jack 9 to top jack 17, and another cable from bottom jack 17 to bottom jack 9. Now the compressor is in between the tape deck track one and the console's tape return one.

While mixing, you could run a track directly to a effects processor just by tapping into it's input's and output's just like above.

You could plug an instrument like a keyboard directly into an effects processor and run it to to mixing board, or to the tape deck.

Get it??? You have all of your input's and output's on a jack where you can access it, and all of your normal connections you need to make are already made for you for a normal tracking or mixing environment. Cool!!!

Geez, I otta right a manual!!!
 
oh...my...god. That is some serious info to digest there! Ive read the whole thing through, and i love the way that it is all explained. All the way through, point, example, evaluate. Just like at English GCSE haha.

Thanks dude! :headbang:
 
Dude, THANKS for all this info and for taking your time to write it down! Awesome stuff for people like me who don't know shit about recording and don't know where to start. Love this forum.
 
Holy crap...I forgot about this thread. I will see if I can pull some more stuff up on my free time and post it. Just remember that there is more then one approach to everything.
 
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