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The Craft Of IEM Mixing
Guidelines that foster quality results and happy artists. -
My Big Stupid Recording Failure
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Using Gain Structure Tailoring To Optimize Overall…
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In Profile: Kevin Margolin & Atomic Professional…
A few years ago, I was given tickets to a concert at the local arena. It was a co-headlining show where one act played, then the other, ending with both acts playing together.
Being the supreme el cheapo, I eagerly accepted the tickets and acted like Romeo when informing my wife that I was taking her out.
We arrived early and naturally I took stock of the audio setup. Each act had its own reinforcement system complete with separate FOH, monitor mix positions, and line arrays du jour.
Admittedly, one act had larger arrays, but like my dad once told me, when fishing, it’s the wiggle that gets the fish, not the size of the worm.
For a few moments I entertained the thought of going down and schmoozing with the front-of-house engineers, but that would have entailed a lot of smooth talking and/or somehow muscling my way past Butch and Bubba (one of which I’m sure had two x chromosomes) to get to the floor. I abandoned the thought and contented myself with enjoying the show to come.
Unfortunately, the audio for act one was a disappointment. In fact, it was horrible.
Act two, on the other hand, sounded fantastic. Why?
Was it because act two had the larger system and thus could put out greater SPL? Nope - I’m too old to fall for that one. When someone declares an immediate and unequivocal improvement in what they’re listening to, I get out my SPL meter and compare the levels. Don’t even try to get that one past me.

What I heard was the difference between running a console at unity and running it without headroom. You can tell the difference just by listening? Yes!
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Good unity mixes sound open, alive, immediate and unrestrained while mixes overdriven in the console sound small, closed, lifeless and harsh.
It has been my observation that when the system is properly set up and aligned, and the sound ain’t so great, the console faders tend to look like Picture A.
Conversely, when it sounds good, the faders look like Picture B. Where the faders are positioned has everything to do with the channel preamp gain setting.

When I first got into sound as a wee lad, one of the earliest questions I had was how to properly set the preamp gain on the console.
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The answer, of course, was to turn the knob until the little red light flashed and then back it down until the flashing stopped. Not. This produced maximum signal-to-noise ratio, but there was absolutely no headroom at the mix bus summing amplifier at unity.
Today, I so wish I could slap that person like they deserve, but that’s what I knew and that’s what I did. My mixes looked like Picture A, and most importantly, sounded like it. Something felt audibly wrong, but I couldn’t find a way to hear outside of that paradigm.
Almost every 'pro' sound tech (guys who do sound on 'known' bands tours) that I've worked along side do this whole 'fader at unity' technique. I ask them all why they do it and the only answer I've heard is, "it's how I was taught". Then every single one of them who have to run monitors from FOH never have enough gain to drive their aux's for monitors. . . THEN the performers get upset when their monitor mixes change constantly as the mixing engineer is tweaking his gains.
After taking Electronics Engineering and having to build A to D converters at school I understand how they work. As a preliminary statement, let me say that I also am a recording engineer and would like to point out that there is a reason people record at high bit depths - it's to achieve a more 'analog' (smooth) waveform. See photo of analog (or high bit depth and sample rate) vs low bit depth/low sample rate/low gained digital waveform here:
http://media.soundonsound.com/sos/sep05/images/forgotten1aquantisation.l.jpg
Quoting the webpage from where I got that picture... "With a higher bit rate, there are more possible amplitude values". Well technically speaking, the larger the waveform going into a A to D converter, the truer the representation of the waveform. If the preamp is turned way down (because you want that channel on your mixing console quiet in the mix), then you may end up with a block like digital square wave (sounding like junk) vs a truer to analog type waveform with a properly gained input to your converter.
To touch back onto my earlier statement about recording at higher bit depths, it's mostly heard in things like reverb tails where near the end of the tail are very small signals... which in the digital realm end up being square waves.
Having said all of that, your theory is flawed... big time. . . at least on digital consoles.
For analog consoles, it's all about signal to noise ratio. EVERY single electronic component introduces noise. So technically you SHOULD keep your signal to noise ratio as awesome as possible... in other words keep your signal as large as possible.
Lastly, if your having troubles with clipping summing stages (like you say in your article), fix yourself. haha Keep an eye on everything, that's your job. If you clip after a compressor, turn your makeup gain down! If you're clipping the stereo buss input, slide all your faders (or group faders) down! Making sure you're not clipping is the first step to mixing!
Sadly, the people who mix with their faders at unity probably will always mix that way. . . and I'll continue to laugh inside at them haha
Please read Matthew 6:5-7.
Jesus doesn't care about the prayers in the megachurches. Jesus doesn't need a huge PA system. Jesus thinks your flashy pastors and preachers are all hypocrites.
Read your book and tell me what I say is wrong.
Some comments. You've "designed" ADCs, so you should have some familiarity with sampling theory. As such, you'd know full well that the picture to which you linked, which shows a stair-step approximation of the digitized waveform, is completely _WRONG_. Hint: a sample represents the amplitude of the signal at the sampling instant. There is no amplitude value between the samples. It could like like this: http://cnx.org/content/m0009/latest/cosine.png
To get back to a continuous-time signal, you need to apply a reconstruction filter, which is what fills in the time between discrete samples. And this is _independent_ of sampling frequency or sample word length ("bit depth" is the wrong term).
Second, the reason for using longer word lengths is to increase resolution at the bottom end. A 24-bit word pushes the quantization noise down below the noise floor of the analog electronics. So turning down the preamp and digitizing a cold signal won't give you that "block like" signal you describe. Reverb tails are _NOT_ square waves. As noted, that's NOT how sampling works.
For BOTH analog and digital consoles, you make a trade-off, which is summing headroom vs signal-to-noise ratio. In both cases, if you trim the inputs hot, in order to mix without overloading a mix bus, you have to pull down the input faders. You can get away with hot input trims when doing a solo-acoustic act on a big console, because of how the console's summing attenuates each input, but when you run all of the inputs hot, you're in trouble.
Please, go back and study your textbooks about sampling theory and come back to the conversation when you actually know what you're talking about. It's clear that Cadwallader has no idea what he's talking about -- don't be like him.