Thursday, February 28, 2013

Sample Rate Conundrum

After yesterday's onslaught of digital audio links I'd love to get back to writing but I've been so absorbed getting back into the minutiae of sample theory that I blew all my free time reading. The thing that's really been sticking in my head this time around was the fact that not only is 44.1/16 basically sufficient for all our audio needs, it's even more capable than I previously thought when handled correctly.

All these studies combined with a vague remembrance of another article I read about high sample rate possibly hurting your audio instead of helping it. I has mostly to do with distortion from  intermodulation that actually has more to do with your amp and speakers than anything in the digital chain. Anyway, here's the article.

Then right before I closed the laptop I remembered one more article that addressed the build quality of the gear. It states pretty simply that a cheap interface at a high sample rate can sound worse than a good one at a lower rate.

And just for good measure, since I'm totally slacking off on actually writing anything, one more article on sample rate that has to do with working in the box with plugins.

So there you go Brethren of the Knob and Fader. Way more reading than you bargained for when you clicked in here today. The fundamentals of digital audio are so interesting, but I'm in over my head when I try to explain it. I hope you're getting something out of this. There's real science behind all the hype. You just really have to dig to get at it.

Wednesday, February 27, 2013

Digital Audio Extravaganza

I don't often just hand out links to other stuff but today I stumbled on such a smorgasbord or digital audio goodness over on r/audioengineering that I just had to share it. Digital audio and sampling theory is deep stuff. What's more is that there's a lot of hogwash floating around out there on top of all that deep, dark theory. The link below brings you not one but three great opportunities to learn and discern what's for real and what's marketing.

Start out with the article on wired. The video takes about 30 minutes and it's one of the best I've seen for getting up to speed on the subject. 
Then there's more reading if you click through. The article debunking some high sample rate mythology is a little hard to find so here it is.

Then if you're not quite reeling from all the input, there's another super archive of info on not just digital audio but all digital media with further links to even more information!

That should keep you busy for a little while Brethren of the Knob and Fader. But if not, then let me take it just one step further and recommend my favorite book on the subject. I read "Principles of Digital Audio" when I was in college in the mid-nineties. It was the text for a studio class that the prof didn't actually expect us to read. But I did and it changed my life. Again it's pretty deep. Getting through some of those chapters on error correction and time code were a chore. The first few chapters alone are worth the price of admission alone and the new edition brings a lot more to the table as there have been some pretty big advances recently.

Tuesday, February 26, 2013

Apparent Loudness

This is a tricky subject. Psycho-acoustics is a deep pond to swim in and I may not have enough air in my tanks to get to the bottom of it, but I'm going to give it a try. The particular area I want to go over today deals with how loud something seems to be. 

You can play an uncompressed sample and then compress it and most people will identify the compressed version as the better sounding one. There can be a lot of reasons for that. A little light compression (with appropriate make up gain) will bring quieter sounds up in a mix and things will sound fuller. The average level increases as well which will make the compressed track sound louder too, even though the peaks aren't going as high as the original. 

So if a little compression is good then more would be better, right? Welcome to the late 1990s and the Loudness Wars. Technology has progressed to where you can just about make a song where the level doesn't stray more than one decibel from a full scale reading for the duration. It even spawned a new term, "hyper-compression". It's really just a continuation of the contest that record companies have been a part of since the 1950s. If your record is a little bit louder than the other records on the radio it will get more notice. But there are better ways.

But there's more than one way to skin a cat. The problem when you go too far with compression is that everything goes flat. Nothing sounds natural. A whispery ballad is the same volume as a brutal dub step drop when they're both coming out of your clock radio in the morning. That's just not natural. What makes things exciting to listen to is dynamic range. The trick is to balance the amount of compression needed to make the mix full with the amount of dynamics that you leave in to keep it interesting.

Take for example a kick drum. Untreated it sounds blah and boring. Start applying compression and it will fatten up. But go too far and it will just get flabby and loose all of its punch. One way to balance is to relax the attack time on the compressor so that the first transient of the beater hitting the drum gets through as a big spike, then the compressor engages and squeezes the resonance out. Or you could split the track and keep one side uncompressed and heavily compress the other one. By blending them together you can keep the interest of the attack and blend in the resonance of the drum.

Another way to create loudness is to compare it to something soft. How many songs have a dramatic pause before things really kick in? And how much more incredible is that break after a measure of silence? Everyone from Led Zeppelin to Skrillex knows it. If you want to make a building look taller, dig a hole next to it.

But tricks aside, with production getting ever denser it's really tough to keep dynamics in a mix. Whether it's a metal track with multiple guitar layers or EDM with a bucket full of crazy synth sounds, eventually you have to get it to come together as a mix and not turn to mush. So let's take it one method at a time.

Stereo width can play a big part. People are nervous about their mixes summing correctly to mono. They should be. But there are plenty of ways to push the pads and subtle rhythms out to the sides to leave room for stuff in the middle that will still work out if a DJ plays it back on a mono system or it gets piped into a department store.

Next is careful management of frequencies. Chances are if you go into a project and tune up each track without listening to the whole thing together it's going to come out awful. Adjustments need to be made while taking into account how they effect the whole mix. Kick drum is a great example. Most people think about 60 to 80 Hz when they talk about kick drum, and probably something up around 4 kHz for that beater sound. But what about 200? Is there room for some 400 in the mix? People spend all kinds of time clearing out the low mids when they're working on guitars and vocals. With good reason, they muddy things up. But if you're not putting anything back in there you're just wasting the space you created. Why not find the resonant frequency of your kick drum, go up a few octaves and enhance some frequencies that will actually play back on laptop speakers?

And lastly, we come back to where we started, dynamic processing. If you're not thinking of your compressors as EQs then you really should try to get your head around it. A comp will make a vocalist sound good if it brings out some nice mids in their voice, but it can also make them sound shrill if the wrong frequencies are emphasized. That's why using side chaining or multi-band compression is a great idea. Make a compressor work on exactly what you want to. It doesn't have to just be a traffic cop watching out for speeders. 

Throw a multi-band compressor on your two mix and work it over. Grab some mid frequencies, shove the ratio right up and drag the threshold way down. Now do it with the lows and the highs. Get so you have a feel how each separate area of your mix responds to it. Even better if you can mute some of the bands while you work on another. Now you can listen to specific frequency segments and figure out what they need. 
 
Is there room in the lows? Or is that area full up? Is the high hat the only thing hanging out above 4k? Maybe you could incorporate some click from the kick drum and make it even louder without giving up much headroom. How do your vocals interact with other things in the mid range? What happens when the vocals stop? There's a whole world in there to explore, section by section.

Well Brethren of the Knob and Fader, I feel like I've been swimming for hours and I'm still nowhere near the bottom of the pond. This is a deep, deep subject and there's a lot more to get in to. Stay tuned till next time and hit us up with your questions and comments.

Thanks to 4Flexx for prompting me to write this. It was good to think through this stuff and get a lot of different ideas together on the same page.

Monday, February 25, 2013

Don't Let A Machine Think For You

This is a concept that's come up before but it seems like it's high time to devote a post to it. Computers have been getting steadily smaller and more capable, as well as the programs that run on them for a long time now. To me it's pretty amazing that an RTA used to be a rack mount piece of gear that had five LEDs per band and that was all the resolution you got. For thousands of dollars you could get a more advanced machine. Now for free you can get a pretty terrific RTA on your phone. You wouldn't want to do laboratory grade research with it but it could help you through ringing out monitors if you were in a hurry.
 
An RTA is a great example of a tool that's overused. A lot of people seem to think that EQing a room until the RTA reads flat is a job well done. But a room that reads perfectly flat on a simple RTA almost never sounds good, for more than one reason. But this isn't an article about room analysis. It's about using tools for their intended purpose. It's also about retaining one of the greatest gifts that humanity possesses. Taste.
 
I think the shortest route might be to explain. I got a plugin for free during a holiday giveaway. It was the One Knob - Louder plugin. The literature said it uses a complex expansion/compression/gain algorithm to make your music louder without messing it up or adding a lot of noise. I tried it out on some dialogue one time when I was in a hurry and it worked great. But after that I shied away from it and stuck to the methods I already knew. Only having one control is a nice idea, but I'm a little leery about a complex process going on behind a GUI that I can't get at.
 
Eventually I sat down with it to figure out what made it tick. I'm not going to sit here and say that I could create identical results with an expander, compressor and some gain, but I could get pretty close and from repeated A/B testing I feel like I have a handle  on what's going on when I twist that knob. It's a pretty darn good plugin. It does just what it says and it's pretty hard to screw up. But it does have its limitations and now I know roughly where they are. I have a great tool that saves me time but I know when it's not the right tool and I have to grab plugins with more control to get what I need out of a track.

The same thing goes for almost anything. Even if it's a more complex plugin or device that gives you lots of control, if you're just grabbing the presets you may not be getting what's best for your project. Sure the kick drum preset on a compressor plugin might work magic but maybe I can do better. I'm at least going to give it a shot when I have a minute and see if I can't improve. Presets are a great way to pick something apart to learn it and often will save you loads of time. But if all you're doing is clicking through presets then you're not an engineer, you're not even really a mixer in my opinion, you're just an appliance operator. The same way someone that microwaving a TV dinner isn't a chef.

I'll leave you with one more example. I bought a Feedback Destroyer once upon a time. I sold it after doing about four gigs with it. At first I thought it was great, that I could toss out my graphic EQs and just let the box worry about it. But after a couple weekends spent basically arguing with the stupid thing about what is feedback and what is, in fact, a guitar solo... I sold that hunk of crap and settled down to the business of worrying about my own EQ again. 

Brethren of the Knob and Fader. You have a wonderful piece of computational equipment between your ears. With that tool alone you are one step ahead of all the technology in the world. You can write cunning programs, you can give them all sorts of information and even make it seem like they're sort of intelligent. But I don't think it's possible to give a machine taste. If they ever do come up with a machine that wonderful I would encourage the inventors to put it to good use, like replacing congressmen with it. Until then Brethren, use machines to your advantage. Use them to save your valuable time. But don't let them think for you.

Sunday, February 24, 2013

Mission Statement

It's been a while since we've laid it out but it seems like a good time to re-post the mission statement. Right from the get go we were never out to write the last word on anything. We don't do tutorials or white papers. The net is full of them. We're just posting ideas, things to get the juices flowing. All the better if people get in on the comments and start some conversation.

We purposely write so that all the bases aren't covered. We assume that you know some things already and that if you don't you'll click over to Wikipedia or another forum and get yourself up to where you can understand what we're saying. That's how we learned.

The podcasts are much the same way. If you're looking for slick production go check out Pensado's Place. It's a top notch, well produced, pleasing to look at affair with top notch guests.  We're sound guys. We get together and start talking shop before our coats are off. Once a week we press record for an hour and post it to the internet. In our own lives our favorite learning experiences have been listening to older, wiser sound guys shootin' the breeze. It's not ever going to be a fact-a-minute educational experience but you might pick up a trick here and there. The old hands among you will be able to laugh along with a battle story or two.

And that, Brethren of the Knob and Fader, is what we're all about. If you want style and polish, we're not it. We're just guys who push faders for a living.

SNR Podcast #37 - 2/24/2013 - Drum Episode #2

We gathered our panel of drummer slash sound guys again this week to pick up where we left off. Last time in Part 1 we covered shells and heads, this time it's cymbals, best practices, micing and anything else we could think of. As always you can check out the YouTube stream right here or use the MP3 link below to stream or save for later.


Friday, February 22, 2013

One Thing After Another

Sometimes life kicks in. Sometimes you get a day off and you expect to be able to sip some coffee, read to the kids, and later on load in a musical for a local high school. Then sometimes you wake up dizzy and nauseous. The kids still want to be read to and you still have that musical to load in.

But we press on. Stay tuned for some good posts coming up. We've had some great requests from readers and listeners and we'll be putting on our thinking caps and doing a little extra research to bring you our best next week.

Thursday, February 21, 2013

Clipping

A question popped up in a forum the other day about clipping and that got the blogging juices flowing. Here's the original question. (the names have been omitted to protect the innocent)

I've read on many forums that clipping is often used in order to get a hot master, but what kind of clipping is this? Are they referring to engineers pushing it hard on analogue gear, or over-compressing/limiting audio in which is being clipped as a result? Are they referring to soft-clipping plugins?

Audio amplifiers, from the tiny ones on chips in a mixer to the big ones on guitar stacks all operate on the same principle. You have a large reservoir of power and you use an audio signal to modulate it. Think of it as a dam holding back water and the audio signal is telling the sluice operator how much to let through at any given moment.

In a good system, the audio coming out looks just like the audio going in except bigger. There are a lot of factors that can cause this to not be true, or as true. Let's assume that good components are in use and we can expect a fairly true output.

If you keep turning up the level of the input, eventually the power supply just won't have enough energy to do what the input is asking it to do. Any requested output that exceeds this limit gets flat topped. The audio looks normal up to the point where the output runs out of gas but any peaks going in above that point are just leveled off at the maximum voltage.

The simple answer is that you just raise the input until you get there but different systems handle clipping in different ways. Solid state amplifiers (transistors) usually run out of gas in a fairly nasty way that adds all sorts of unpleasant harmonics to the signal. Tubes, and tape machines have a much smoother transition which lends a much more pleasing set of harmonics. Transformers saturate in a nice way too which is one of the reasons so many people like big, old analog consoles. Digital on the other hand is very cut and dried. Once you run out of numbers your signal is instantly flat topped, no transition, lots of unpleasantness.

There's a whole lot more to this that you need to understand before you set out to clip something to make it louder. There are a lot safer ways like compression and limiting with the application of make up gain to get your stuff loud. In fact about the only time I can think of where it's common to clip is with the amps driving the subs in a large format audio system. But even that is starting to become frowned upon due to the falling cost of amplification and the implications of the harmonics generated.

One last thing. If you're using plugins, many are designed to simulate analog devices which are capable of producing nice sounds when driven into saturation (clipping). In that case it's OK to clip because it's a simulation and not an actual digital clip.

Then another redditor jumped in with another doosie of a comment which I shall now share with you as well.

To add upon [that elegant explanation], psychoacoustics (the science of how sound is perceived) dictates that we are more sensitive in the midrange, about where the human voice sits. Those curves are essentially the inverse of our hearing sensitivity: they display the loudness needed for various frequencies to be perceived at an equal loudness, hence the name. If you have sounds outside of the most sensitive range you can turn them up, but eventually they start to fight each other. You can increase the perceived loudness of these sounds by having them generate harmonics in the range of our greatest sensitivity and we also tend to perceive the additional harmonics as part of the original signal because they are a type of correlated noise. One way of doing this is by clipping the signal: the way the circuit behaves when it clips includes the generation of harmonics, some types of circuits tend to generate higher-order harmonics (solid state) and some tend to generate lower-order harmonics (tubes). As an extreme example, this is what overdrive and distortion effects do to varying degrees and the effect can be measured as THD (total harmonic distortion). As a less extreme example, this is one aspect of what devices like the BBE Sonic Maximizer or the Aphex Aural Exciter do.

This of course is aside from the fact that if you're clipping a signal, the parts that aren't clipping are being brought up in level along with the rest of the signal.

Just one final note. The audio forums over on Reddit are some of the best around. The ones at r/audioengineering and r/livesound in particular are flame and troll free for the most part. Just people asking questions and exchanging ideas. You should check it out.
 

Wednesday, February 20, 2013

Multiband Compressors - Resist The Urge

For well over a decade now manufacturers have been selling multiband compressors as a panacea for your mix. Hardware units like the TC Finalizer and later on plugins. The problem is that while judicious use can yield spectacular results, it's difficult to resist temptation. Compression sounds good right? So why wouldn't more compression sound better?

Because if you can suck the life out of the project by compressing the full spectrum, you can multiply the ill effects by doing it four times over.

Now that said, being able to treat the lows like the kick and the bass separately from mids and highs like vocals and guitars can be a godsend. But it's so easy to tip over the edge and just murder a mix, particularly when the makeup gain is set to auto.  Every little bit you pull the threshold down raises the output gain up by an amount determined by the ratio of the compressor (usually). You can get a mix jacked all to heck before the compressors even start to kick in.

Rather than describe in horrifying detail how to kill a mix, let's get on to how to use one of these things the right way.

First of all, they can be handy when used a little bit differently than their intended purpose.  If I'm moving quickly in a session I find that it's faster to slap a multiband compressor on a track as an insert plugin, turn off three of the bands and just use one as a de-esser. For me it's quicker than setting up a side chain and I find that I have better control than with some of the de-esser plugins. It's also possible to use it more like an EQ, treating just the low mids of a guitar track for instance.

That brings me to going about setting up a multiband on a master bus. The first thing to do is carefully determine how wide the bands will be. Treating the lows below 100 Hz might work great on a death metal track but totally murder a jazz track. The same goes for the mids, if not more so. The safest bet is to try to divide it up to treat instruments and vocals separately.

The next thing to do is turn off the auto setting on the makeup gain. It's never a good idea to let a machine do your thinking for you so try and decide for yourself how much gain to put back on after you've finished messing with the rest of the settings. 

If your compressor lets you mute the bands you're not working on that will probably help get things dialed in as far as threshold and ratio. Just make sure you keep switching back to listen to the whole thing once in a while. Remember that everything has to fit together when you're done. Keep in mind that the attack and release times will probably need to be different for each band. What works to keep transients in the kick while fattening it up might not work for the keys and guitars up higher.

At this point a delicate touch is all you need. Strapping a limiter on the two mix and slamming the daylights out of it won't work for every style of music. The same goes but more so when you're doing it three, four, or more times. Not that you should be mixing primarily with your eyes, but if you're seeing six dB of reduction in each band you've probably overdone it. Think of this as a chance to make each section of the spectrum dance a little on its own, while helping to tie the whole mix together.

Once all the settings are dialed in then you can go back and start to add the makeup gain. Just apply carefully and keep an eye on the meters. You may even want to have a regular limiter down stream just as a safety measure. 
 
Just take it easy. That's the key. Ignore the presets. Ignore the urge to keep squeezing and squeezing.  It's not getting better. It's just not. Take deep breaths. Just tickle the settings. Nudge them. Do it nicely and your tracks can come alive.

Tuesday, February 19, 2013

Phase Issues

I've been in a few conversations lately where we wind up talking about fixing phasing issues with drum mics. There's two ways to go about it. You can either go in the live room and move the mics, or you can "flip the phase" which really means hit the polarity button on the console or DAW. I'm going to be very clear which one I mean and you'll see why in a minute.

First of all, everyone says "flip phase" because it rolls off the tongue better than "reverse the polarity". No worries. We're all big kids here and we know what the deal is. But I'll be careful to say exactly what I mean though so there's no confusion about what we're talking about.

Let's start with what the difference is between the two techniques. Moving the mics subtly changes the arrival times of a sound to each mic. Two mics on the same signal will make some frequencies combine additively and others combine subtractively. What you wind up with is a comb filter effect and you just search around until you find positions that produce a comb that emphasizes frequencies that are pleasing. Inverting the polarity of one of the mics is an electrical process that doesn't change the arrival times at all.
Because you're dealing with two mics on a single source, you're dealing with time shift, that's phase. Pressing the polarity button is like going after the problem with a pipe wrench when what you really need is a jeweler's screwdriver. Yet we see people going about getting drum sounds all the time poking the polarity button and thinking they're getting to the root of the issue.

It's a complex one though. The simple answer for the longest time has been to just move the mics. But there's another tool available to people working in a DAW and that's micro delay. That means that if you really like where each mic is sitting individually, you can mess with how they interact with each other by applying tiny amounts of delay. Each foot is roughly equal to a millisecond so we're talking about getting into decimal places here.

I'm not going to sit here and write out the whole process for you because I'm hoping you're starting to see what I'm talking about. This doesn't just apply to the Brethren of the Knob and Fader who inhabit studios either. It's becoming common practice to take direct lines not just from the bass but also from guitars on stage and use micro delay to precisely line up that signal with the mics. In the age of the digital console and line array it's possible to get things razor sharp.

Monday, February 18, 2013

It's Been A Year

It's hard to believe that SNR has been a thing for exactly one year. The original mission was to have a post a day on the topic of pro audio. We fell about sixty short of that goal but after about 250 we figured that quality should come before quantity and decided it would be OK to miss a day here and there.

Of all the things we've done this year to promote the blog and get people talking (not about us but about audio) going on Reddit has been by far the best. The forums at r/livesound and r/audioengineering are two of the best places to go for civilized discussion of our craft anywhere on the internet. Other forums might be bigger or have fancy sponsors, but Reddit is where the gentlemen hang out. Hats off to all the redditors who have contributed ideas and feedback.

Our podcasts have been taking off a little more slowly and we realize that not everyone is going to want to listen to us banter about audio for an hour at a stretch. (You should though cause we're funny as heck sometimes.) What has been catching on though is the mini podcasts. That's where we take eight to twelve minutes and go over a topic with some audio examples and sometimes even have raw tracks for you to download and play with. You can find them on the same page.

We're also on Facebook and Twitter. There's links over on the right side of the blog. If you've got questions, ideas for posts or even corrections for stuff we said wrong, those are the places to get in touch with us. We've already made some great friends on both sides of the Atlantic and we'd love to make some more. 

So enough anniversary gushing. I'll just throw out a little audio tidbit so today's not a total loss if you were coming here trying to learn something.

"It's not a real power amp if you can't arc weld with it."
                                       - Joe Tall

Sunday, February 17, 2013

SNR Podcast #36 - 2/17/2013 - Drum Episode #1

That's right folks, the title indicates that there was so much drum goodness this time around that we didn't even begin to get to it all.  This week your hosts Jon Dayton and Anthony Kosobucki got together with new contributor Brandon Kapral and Steve Kowalcyk who's been on board since the early days. With two guests who are both drummers and sound guys we knew it had to be a drum show. Starting out with shells and moving on to heads, hardware and tuning, time was up before they cymbals came out. Tune in next week when we'll pick up where we left off and then move on to what drummers should do to make sound guys happy and vice versa.

As always you can check out the YouTube version right here or use the MP3 link below to stream or save for later. Happy listening and keep an eye on those PTMs! (Precious Tone Molecules™) Speaking of which, see if you can hear the difference between Anth and Brandon on the Focusrite 2i2 and Jon on the ART DualPre. Steve was coming in on Skype so some allowances have to be made.


Skype Mic

Thursday, February 14, 2013

Star Grounding

This post is part of a series on eliminating noise from audio systems. Click Here to see them all. 

I've got one last trick for beating hum and noise and this comes in particularly handy for guitar players but it's just as important to keep in mind for other applications sometimes too. Everyone knows that grounding is important. Unfortunately those paths for electricity to exit the building safely in case of a "leak" aren't always our friends when trying to keep noise out of audio circuits. That can lead to people lifting the safety ground on equipment.

Let's be clear. It's not just stupid, it's straight up illegal in some instances. So just don't do it. Ever. 

Lifting the ground on a direct box or other audio path is the way to go when you're trying to get rid of common mode noise (ground loop hum). But in a system that's not balanced audio, like a guitar rig, you can't lift the ground and still have signal. It would seem that there's nothing left to be done, but there is.

When you plug in a guitar rig, on stage or in the studio it should be on the same "leg" of electricity as the rest of the audio gear like we talked about in the last post. That alone can kill off a ton of noise. But if the amp and say, the pedal board are plugged into the same circuit but from different outlets, the two devices can be at different ground potentials and that's when Buzz Lightyear shows up as an uninvited guest. 

So here's your method. Run a power strip or other splitter from the point where you're plugging in. Plug the amp into that and then plug an extension cord for the pedal board on that same power strip. What you end up with if you do this for every series of devices in the setup will look like a star if you draw it out. Panel to outlet, outlet to first device, first device to second device. Even using two outlets on the same circuit won't be as clean because you're going panel to first outlet (where the amp is), on to second outlet (where the pedals are) and then you've got a loop once you introduce an unbalanced guitar cord into the mix.

But wait. Aren't the two examples the same? Just taking a different route to the same destination? Yes, that's the key. Physical location. If the power and the signal for the pedal board are traveling roughly the same path through the room they'll be subject to roughly the same interference an have closer to identical ground potential when they arrive. A guitar cable running across a stage and an electrical circuit running through a wall in a conduit are having different experiences before they arrive at that same destination.

I know it might sound complex or like it might just be one more of those voodoo superstitions that abound in this industry but it works. I've had guitar players at their wits end, especially guys with single coil pickups go to this method and it has saved the day time and time again. Even with all the hand built, triple shielded, impedance matched, tone molecule preserving cables, just plugging in the power the right way can make a world of difference.

So there you go Brethren of the Knob and Fader. This concludes our impromptu lesson on how to kill noise on stage and in the studio. Go forth and prosper (and no humming on the way out!).

Wednesday, February 13, 2013

Which Leg?

This post is part of a series on eliminating noise from audio systems. Click Here to see them all.  
 
In the continuing effort to help the Brethren of the Knob and Fader remove all traces of hum from their systems we address electricity today. Let's jump right in and talk about how the power that comes into your place is delivered.

There are two kinds of power commonly seen (and a few less commonly seen but we'll leave grounded B-phase a.k.a. "widowmaker delta" to the professional electricians). Three phase is common in commercial buildings. In those cases there are three hot conductors into the panel and three bus bars that distribute power to the branch circuits. Each row of breakers is on the next leg as you move down the breaker panel. The other kind, seen in residential service is called "split phase" and uses a center tapped transformer to take one of the three phases running down the street and splitting it to power two buses.

The difference between three phase and split phase isn't much if you just consider voltage going to standard outlets. The voltage between any of the phases and the neutral conductor is 120. When you start to compare voltages between phases you'll find that in your house you see 220 volts between the two legs and in commercial three phase you'll see 277 volts. Without getting too deep into it, the legs in a three phase setup have the 60 Hz waveform 120° apart from each other in phase. With split phase the two legs are of opposite polarity but with sine waves that's the same as being 180° out of phase.

OK. Enough with the science. Here's how you use the knowledge of branch circuits to quiet down your audio. Audio equipment will benefit from having everything in the system on the same leg of the panel. That means in a residential panel using outlets that are on the first, third, fifth, etc rows of breakers. In a commercial setting it's the first, fourth, seventh, etc. If your equipment is all on the same leg or phase you'll have a lot more immunity from hum caused by other devices like dimmers and appliances. 

A lot of care is taken to work this stuff out when power is set up for venues and studios. You can even rearrange the breakers in your house to make sure the washing machine and refrigerator are on the opposite leg from your bedroom so you can get cleaner tracks. But what if you don't have access to the panel or it's poorly labeled?

You can often tell if two outlets are on the same leg just by checking the voltage. Because the load on an electrical system is seldom balanced, the voltages vary between legs or phases. Setting up a home studio you have the time to sort things out but loading your bar rig into an unfamiliar venue you need to be quick on the draw, so keep your multimeter handy.

One way to solve the problem though is to just keep extra cable handy. If you're running a guitar rig in the other bedroom or trying to power up the mix position in an electrically noisy room, just bringing power from the control room or stage to the other gear can quiet things down considerably. While the extra length of power cable can be another way for noise into the system, keeping things all on the same leg will do more to reduce noise than gets in on the extra wire. 

That's all for today junior electricians. Tune in tomorrow as we wrap up this impromptu series on noise reduction with a quick lesson on star grounding!

Tuesday, February 12, 2013

Balanced Connections, Always.

This post is part of a series on eliminating noise from audio systems. Click Here to see them all. 

Yesterday we talked about the misconceptions people have about what a power conditioner can do to get noise out of an audio system. Today we're going to look at something that's a lot more effective. Balanced connections.

In short, they use two paths for the same signal. One of the paths inverts the polarity. When those signals are combined at the input of a mixer or other piece of gear, only a signal that has a counterpart on the other conductor gets in and noise gets shut out. The science is a little deep so I'm going to skip it for the moment. Feel free to go look it up though.

The two most common kind of balanced signal connectors are the familiar XLR three pin seen on mics and the TRS 1/4" connector. The XLR has the ground on pin one, the signal on pin two and the inverted signal on pin three. With the 1/4 inch connector the tip is the signal, the ring is the inverted signal and the sleeve is the ground.

Just by making sure you only used balanced connections in your setup you can virtually eliminate noise. Out on gigs I've been in lousy situations where this has saved my life. I've had my amp racks and snake head literally on top of a pile of old SCR dimmers and had very little noise in the system.
Once everything is balanced, you then have the option of lifting the ground on an input if you do still get noise coming in. Direct boxes for electronic instruments are a great case for this. By removing the shield connection at the instrument, the cable is still draining interference off at the mixer end but there is no direct connection between the chassis of the two devices which could be hundreds of feet apart. 

One other thing that will help greatly is making sure all the audio equipment is on the same leg of the power in the building. In a lot of homes there will only be one circuit per room so it's easy. If you decide to use another room for an iso booth you should run extension cords in there to make sure any gear you plug in shares the same power. One easy way to tell if two outlets are on the same leg (even if they're different circuits) is to simply take a voltage reading. If the load is unbalanced in the building (which is usually the case) the two legs will differ by a few volts.

But all that is another discussion entirely Brethren of the Knob and Fader. So check back tomorrow.

Monday, February 11, 2013

Power Conditioners

This post is part of a series on eliminating noise from audio systems. Click Here to see them all.

A lot of people experiencing noise in their audio think to go for a power conditioner first. Unfortunately it's often money wasted as power conditioners only address a few specific issues. They don't do a thing to fight the biggest cause of hum on the stage or in the studio. Ground loops. We'll cover that in the next post. Right now it's down to what a power conditioner can actually do for you.

Your garden variety rack mount power conditioner is basically just a power strip with rack ears. There's a couple features that are usually better implemented than what you'll find in the more expensive pieces of office equipment. Rack gear might have a voltage meter or a nice set of lights but that's just window dressing. A fuse or more likely a circuit breaker or thermal overload will round out the basic features.

What actually counts is filtering and the ability to deal with power surges. The filtering on most units is a simple low pass filter on the hot and neutral connectors right after they come in to the unit. That's right, simple audio technology to filter harmonics from the power. It's a passive filter with all the limitations that any other passive filter will have. Some are better than others. Compared to other issues you can have though, harmonics in the power can be pretty hard to hear in finished audio unless you have a severe issue.

The only other technology is the surge protection. This is usually just a couple MOVs across the terminals of the outlets. An MOV is a metal oxide varistor. It's a device who's internal resistance increases as the voltage across it increases. When a big surge comes down the line the resistance skyrockets in the MOVs, causing them to absorb the spike and produce heat. Many are single use depending on the severity of the spike so there are a lot of racks out there that are basically unprotected after they do their job once. One feature that office surge protectors have is a circuit that indicates when protection is present. Once the MOVs give up the ghost the light goes off and you know it's time to replace.

Some more advanced units will have over and under voltage protection. This is usually some sort of indication when the power slides out of the ideal range and full shutdown if it drifts further. These days voltage conditions are less of an issue as switch mode power supplies can accept quite a wide range of input voltages and still deliver clean, stable DC to the gear.

A UPS (uninterruptible power supply) can do a little more for you, although basic models might do more harm than good. These devices have internal battery packs that run an inverter when mains power is interrupted. A cheap inverter is doing nothing more than passing a 60 Hz square wave. This can really do a number on motors and electronics unless they're specifically rated for inverter duty. Better inverters will supply a smoother sine wave. This technology is constantly coming down in price and even relatively inexpensive inverters can be found with this feature.

The best of the best though are dual conversion setups. This is like a UPS but runs continuously. AC power comes in and is rectified to DC which charges the battery bank, but also continuously runs a very good inverter. Many produce better power than you can get from the utility company and they do it continuously, not just during power outages. These are commonly seen in phone and network rooms where sensitive equipment has to run 24/7. They're expensive though so this type of gear isn't often seen outside of the ritziest studios.

So that's the low down on power conditioners Brethren of the Knob and fader. Check back tomorrow for the skinny on eliminating the rest of the hum from your rig or studio.

Sunday, February 10, 2013

SNR Podcast #35 - 2/10/2013 - Teaser Episode

This week Jon Dayton and Anthony Kosobucki go over a couple projects that are coming up and will likely feature in future posts and podcasts. In addition there will be some new guests to bring you some fresh perspective. There's some thoughts on vintage plugins too. As always you can check out the YouTube version right here or use the MP3 link below to stream or save for later.


  • SNR Podcast #35 - 2/10/2013 - Jon Dayton & Anth Kosobucki tease some ideas for future posts and podcasts and toss in a few thoughts on vintage plugins.

Thursday, February 7, 2013

Authenticity in the Studio

SNR would like to welcom Brandon Kapral to the fold as our newest contributor. He'll be leading off with a multi-part series on a studio enveavor to give some insight to those who have never done it. To keep up on the series just look up Project Authenticity on the Topics page. 

 In order to explain the reasoning behind my first post here at SNR, I first must give you all some background. I have been a fellow audio professional in live and studio settings for more than ten years and have been a drummer for twenty-five (professionally for three years before I decided that it wasn't for me).   Currently, I work for two production companies in Western and Central New York, go to school for Electrical Engineering, and play drums for a band called The Carolina Gentlemen - of which you may have heard about from my guitar player, SNR guru Anth Kosobucki.  This will be the first in a series of posts that will document a cool, if not slightly insane, idea for a recording session.   

I have worked on dozens, if not hundreds, of studio projects, as an engineer, a producer, or an artist, and as we all know, these projects are quite varied in their construction. As engineers, personal projects are sometimes the most difficult to nail down.  We know (think) that we have the talent (stupidity) to create that "perfect" product.  It's almost always something that we do in our "spare time" and without a ton of outside influence from other engineers, or at the very least, no other sets of ears to give their two cents.  Sure, it sounds great - no deadlines, no money concerns, and most of all, no pesky bosses breathing down our necks.  Really, though, all of these things can sometimes come together to create a storm of insecurities and roadblocks.  Without an outside perspective, we can so easily get lost in our own musical and technical thoughts.

So where is all of this going, you may ask?  Well, for the last year-and-change, Anth and I have been recording an album for The Carolina Gentlemen.  Our singer, Wes Walters, has written some truly great songs that deserve the best that we can give them as far as production and performance are concerned.  We have spent countless hours experimenting with instruments, mics and mic positions, tracking, dubbing,  pre-mixing, editing, mixing, re-mixing, etc.  A year is a very long time and we put a ton of work into this thing.  There were a few breaks as our members got married and had babies, but for the most part, we worked on it at least once a month and when we got on a roll, it would be 5-10 days per month.

And now we are going to scrap the whole thing.  All 18 tunes.  And it is all my idea (fault).

The proverbial light bulb came on as I was listening to Marc Maron's WTF podcast featuring an interview with Dave Grohl.  Marc likes to present topics for his podcasts to help with continuity, and the subject of the Dave Grohl edition was authenticity.  Mr. Grohl has a new movie/album/tour out called Sound City.  The subject of the movie is its' namesake, LA's former Sound City Studios, and more specifically, the recording console that used to be there.  This custom Neve 8028, Grohl believes, is the magic tool that produced a huge amount of top records from the 70's, 80's and 90's, from Fleetwood Mac's Rumours to Nirvana's Nevermind.  He so much believes in that magic that he bought it and transported it to his personal studio. What Grohl loves so much about this console, he explains, is that it is a piece of gear that is incredibly unique and authentic.  It was designed, fabricated and assembled by hand.  Some guy had his deft paws on a soldering iron for a long time to create the console that led to so many musical classics.  

One of the things that he mentioned on the tech side was that the monitoring section of the console is unique to that particular console.  According to him it is very simplified, making the signal chain shorter and the sound more "pure".  I am paraphrasing here, but he says something along the lines of 'What comes out of a singer's mouth goes through the mic and directly to your f*ckin' ears, so you better be good!'  They didn't ever record thinking that they would "fix it in 'Tools".  He told a story about his experience recording the first Foo Fighters album, The Colour and the Shape on which he sang, played guitar and re-tracked a previous drummers parts to make them his own. His producer, Gil Norton, made him play his parts hundreds of times until he "got it right".  They were tracking to tape, and while it was possible to edit tape, Norton wouldn't do it.  He would make Grohl come in from "noon until midnight" to track one drum part.  That is truly authentic, and you can hear it if you listen to that record.  It happens to be one of my favorites.    

All of his stories and ideas got me thinking - what if we were doing it all wrong?  We had spent so many hours tracking each instrument individually and playing with technology to improve it or edit the performances, yet I felt like we were always chasing something in the mix to make it better.  Each tune had it positives, but I kept hearing little things that made me unhappy.  When I talked with Wes, he posed the question - "If you look back on this record in 5 years, would you be proud of it?"  That is exactly the question that I needed to hear to, as the Crüe would say, kickstart my heart.  So I talked to the guys.  We're gonna can it and start anew.  And do it more authentically this time.

Anth , Jon and I have a lot of ideas for the tech side of things, but the overall idea is going to be to track to Pro Tools (because it is what we have), but use it as a tape machine.  If I have to play a drum part 100 times to get it right, so be it.  We are going to track as much as a group as possible.  No edits. Our live shows and rehearsals have a lot of heart that the previous tracking sessions lacked. We are going to try to capture that here.  This will necessitate our time and attention, but really, isn't that what it deserves?  Why do it half-heartedly - who will that actually benefit in the long-run?

We begin setting up gear in Anth's studio tomorrow morning - I will try to remember to photo-document this thing as we progress.  As you can tell, I am excited!  I think that we all have a new fire lit under our butts that will hopefully translate into a great record.  I hope you'll stay with me as Anth, Jon, and I venture into this crazy thing.

Wednesday, February 6, 2013

The Old Four Track

Tascam Portastudio
I don't know if there's any technical merit in revisiting the days when the four track cassette machine was the sole means of multitrack recording available to the aspiring teenage musician. There probably aren't but I'm feeling nostalgic so that's where we're going today.

How many of us spent our high school days banging out cover tunes and maybe even daring to write an original out in the garage or down in the basement. At some point the dreams of rock and roll fame got to be too much and we got our hands on a four track tape machine. There was quite a bit of satisfaction to be had when a month's pay was finally shelled out on that second hand Tascam or Fostex. You felt like you were in the big leagues.

Then it was time to get down to business and try to sort out how to cram the whole band on there. Drum track, bass track, guitar track vocal track, record the whole band live and be done with it? Or you could go the route of using the rehearsal PA to get a stereo mix going of the whole band and coming back later to overdub vocals and solos like the big boys. If you got really creative you could use all the mics you owned to do a huge drum session and bounce that down to two, then keep adding tracks and bouncing down. You could wind up with a decently wide stereo mix at the expense of several generations of crappy dubs.

The problem was that four tracks on a quarter inch of tape was just too much in too little space. There literally wasn't physically room on there for any serious amount of dynamic range. And as for that "tape warmth" that people ooh and ahh about these days, there just wasn't any on those machines. The inputs sucked, the EQs sucked, the heads sucked, and even if you spent all your paper route money on classy CrO² high bias tapes, you still just had a big pile of hiss that if you were lucky something mildly intelligible came out of.

Which brings to light the second problem. Nobody making music on one of these things had the slightest idea what dynamic processing was. If you played too loud it clipped out and if you played too soft it all got lost under the high hat. Oh, the misery that a single channel of dbx compression could have alleviated!

The really interesting thing is how the cost of home recording has diminished over the years. When TEAC released the first Portastudio in 1979 it cost $1200 (in 1979 dollars!). In the early nineties you could get a mini disc based eight track machine for that kind of money. Now days, for $1200 you can get a pretty bangin' laptop, a student edition of Pro Tools, a USB interface with pretty decent pres in it, and still have a couple bucks left over for a mic or two.

There you go Brethren of the Knob and Fader. An excuse to look back and sigh (or shudder) at the technology of days gone by.  Let's hear it for progress!

Tuesday, February 5, 2013

Drums Galore and Pro Tools 11

I'm kind of taking the lazy route today but once in a while there's good information to pass on and so I just leave it to the original source.

The first bit of goodness I have for you today is a drum website from a friend of a friend, Chris Brush. I got turned on to it when a friend linked up to a humorous video about what drum fills are acceptable, questionable, and downright awful for session drumming. The rest of the website includes brief video samples of different setups which are enlightening in their own right. Beyond that, you can download the multitracks to play with and that right there Brethren of the Knob and Fader is pure gold. In addition to all that it's a chance to see how a musician has uniquely packaged himself. I love the idea of a drum menu. Check it out!




Next up is a selection from Dave Pensado's vlog. He had Bobbi Lombardi form Avid on as a guest a while back and you really need to pay attention to this guy! He talks fast and covers a lot of ground but it's all worth a listen. They work through 32 bit vs 64 bit, swing by floating point, and then get into the development of Pro Tools 10 and beyond into the work that's going into PT 11. The guest segment starts at 12 minutes.

Monday, February 4, 2013

SNR Mini Podcast - Guitar Mic Trick

This little trick is one we've talked about before on the blog and the podcast. I'm pleased to finally be able to offer you some audio examples of it in use. There are lots of techniques for using multiple mics on a guitar cabinet but most of them involve using a close mic and another mic anywhere from a few feet away to across the room. The goal in these cases is to include the room sound. 

What's done a lot less often is using two close mics. I stumbled on this trick the day I got my first Sennheiser e609s. I threw them out on stage along with some SM 57s and planned to A/B them on each guitarist as the night went on and just pick the one that sounded best for each player. Five seconds into the first sound check I found that I had a fat mic and a skinny mic and I could create guitar Nirvana just by balancing the two channels against each other. I never looked back.

The principle in effect here is phase cancellation caused by the difference in position of the two mics and the slightly different frequency responses. It causes comb filtering and you control how much by changing the gain on each mic. It makes mic positioning a lot less important which is a nice advantage on a festival stage. if you get an inexperienced opener with screechy tone, you just push up the fat mic and ride the skinny mic down. With more experienced players it lets you react to the tonal changes they create. You can make the fat tones a little fatter or mellow out the solos to get them to fit perfectly in the mix.

Another advantage is that in a dense mix, you can make a solo cut through without making it louder. I find that I generally ride with the fat mic a little higher than the skinny mic. When a solo comes up I just swap the positions of the faders and the notes stand up in the mix without taking anyone's head off. 

So there you have it Brethren of the Knob and Fader. With the addition of a second mic to your guitar inputs you open up a world of possibilities. Advanced users can even try micing the back of a cab. You'll find anything from additional corpulent fatness on a Fender open back combo, to bone crushing bass notes coming off the back of a Marshall half stack. Use with caution though, it can get ugly and out of control in a hurry.

Below you can catch Anth's mini podcast with the MP3 link to stream or save for later. Below that you can find the three sample tracks so you can put up a session in your DAW and see for yourself how to use this trick.

Sunday, February 3, 2013

SNR Podcast #34 - 2/3/2013 - Internet Success, Sax and More

This week Jon Dayton and Anthony Kosobucki flip out over a post that went viral on Reddit, talk over a sax recording session, and move on to a host of other things. Plans are being made for more mini podcasts so get your requests in on the Facebook page. As always you can check out the YouTube stream right here or use the MP3 link below to stream or save for later.



Friday, February 1, 2013

Silent Mic Switching

There are a lot of people, some even with a good deal of experience, who don't know this little tidbit about how the switches on microphones work. I had to work this out the hard way as a young sound guy with no internet. Years later I finally learned that just breaking the continuity isn't what shuts the mic off. It's actually something sort of counter intuitive.

On a balanced mic cable you have one conductor as a shield and two conductors carying the same signal but with opposite polarity. If you stick a switch anywhere in there that just breaks the signal, you're going to get a pop. If the phantom power is on you're going to get a really nasty pop. And if the phantom is on and you only break one conductor but not the other you're in danger of ruining a microphone.

Instead what you have to do is make those two conductors meet up. Identical signals of opposite polarity will cancel perfectly when summed. The beauty of this method is that because phantom power is delivered as +48 volts DC on each of those two conductors, they're both at the same voltage and therefore with respect to each other are at zero volts. Just let that last one go by you if you didn't get it. Phantom power is another lesson.

So how do you do this with a switch? Let me make the simplest possible diagram. I'm not even going to open MSPaint for this one, we're going old school ASCII art. (Just let that one go by too if you're too young to remember newsgroups.) On a double pole, double throw switch you've got six terminals that you can hook wires to. The two in the center of each side are usually the common, and the ones at either ends can be thought of as different outputs. When the switch is down, the center terminals are both connected to the bottom pins on that side (but kept separate, that's why it's called a double pole switch) and with the switch up the signal goes to the top pair of pins.

So how do you get those two separate but opposite signals to meet in the middle when you want the mic off (muted actually) and still keep them separate when you want it on? You only use two of the terminals on the same side of the switch. You don't even need to cut the wires from the diaphragm of the mic, just bare a little spot and solder each one to one side of the switch. When the switch is down, the signal doesn't pass through the switch at all, just passes right through the top two pins and on its way out to the mixer. Like this. The "C" is the contact off the switch. Technically wire two is electrically connected to the bottom terminal but since it's not connected to anything there's no issue.
            |     |--wire1
             |     |--wire 2
| C |

When you want the mic muted, you move the switch up and the contact bridges the top two terminals and the two wires are shorted together. Normal and reversed polarity signals cancel each other out, phantom power remains balanced and your mic goes quiet. 
            | C |--wire1
             |     |--wire 2
|     |
 
There you have it Brethren of the Knob and Fader. Simple electronics at work for you on the gig.