Trying Something New

August 11, 2009

In recording and pro-audio it’s all too easy to fall into a rut and never try new things. You have found a variety of mics that you like for certain instruments, particular pieces of outboard gear you always fall back on, and mixing techniques that you constantly use.

There are a number of reasons for that, of course. The biggest is probably time, which often equals money. In live situations, there usually simply isn’t time to experiment, unless you are fortunate enough to work with an act big enough where you can have technical rehearsals and full-blown dress rehearsals. In the studio, you are most likely working with a band who is under very tight budget constraints, and that is not conducive to being able to spend time trying different things.

When you have the time, though, it can be extremely rewarding, and can result in some stunning results.

There is also a trend started by some “how to” books to sort of ‘mix-by-numbers.’ They tell you that the kick drum should be at about -3VU on the stereo bus meters, and the lead vocals should be -5VU, or some such thing. Nonsense. Only your ears can tell you how loud something needs to be in the mix. There are some shortcuts you can sometimes take. For example, if you’ve mix a particular band often enough, you usually have a pretty good idea of where things are going to be panned, so it wouldn’t hurt to set the pan pots where you think they’ll end up. But, don’t be afraid to play around with those while you are mixing. You just might find a location for something in the stereo spread that works better for that particular mix. There are a lot of very stunning effects that can be accomplished with phase, delay, and panning, and you will never stumble across any of them if you don’t take the time to try something new.

Sometimes your mic selection and placement has gotten to be such a habit, that you don’t even consider something really different. The absolute best cranked Les Paul/Marshall tone I’ve ever heard was accomplished by a mic selection and mic placement that I NEVER would have tried. But someone took the time to try it, and found that it really worked.

Another, often ignored reason is that digital equipment makes it difficult, if not impossible, to do much “creative patching.” In the old days, you had a patch bay, with a number of mult points, and you could patch anything to anywhere, combining with other things or splitting the signal along the way. With a digital multi-effect unit, this is simply not possible. Most digital mixers and digital audio workstations impose rather rigid signal flow ideas. Beyond that, it is simply not possible to really explore parameters of the equipment that weren’t programmed into it by the developers.

But whatever your situation, try different things. The next time you are tracking a guitar, use your regular mic and placement, but also set up a completely different mic, and use a different placement. Record it on a separate track so that if you don’t like it, you haven’t cost anything, and compare the two. You might just find that it offers something you didn’t expect, and maybe something you can use alongside the other track in the mix.


Digital Doo-Dads

June 23, 2009

Every time I listen to a digital gadget at the local music store, I am overwhelmed by the mushiness of the patches. Or, maybe a better way to put it is that I am underwhelmed with the usefulness of the gadget. If you assume that the factory patches are the best that it can sound, you probably won’t buy it. Be assured that these factory patches aren’t representative, and are probably almost the worst that it can sound.

Plug a guitar into the latest whiz-bang modeler, select any of the factory patches, and you will assaulted by the combination of every effect known to mankind. All at once. With NO dynamics. How can the people who make these things go to all the trouble to “model” all of those vintage effects pedals and amps, and then have NO CLUE as to what they should sound like when you use them?

This holds true on the guitar modelers as well as digital synthesizers. I’m not saying that they CAN’T sound good. Some of them are capable of sounding very good indeed. It’s just that, if you expect to unpack the thing, select a factory preset, and have it sound good in your band’s mix, you are in for a big disappointment.

There are examples of this in all of them, but one standard factory preset on guitar modelers that seems to transcend brand name is the heavily distorted AND highly compressed patch, with a ton of fizzy distortion, always a lot of chorus, some reverb, and a really scooped EQ. It might sound fun in a music store through some little transistor amp played at 60dB, but it’s not going to work through a 4×12 cabinet, crunched tube amp, played at 110dB, and it’s CERTAINLY not going to fit in the mix. Because of all of the extra compression and time-based effects such as chorus and flanging they always add, there will be NO articulation and dynamics.

But, if you build your own patches from scratch, you can come up with some good sounds. Don’t expect them to sound exactly like your guitar through a cranked Marshall with a pair of 4×12 cabinets, for example, but you can create some patches that sound good in their own right, are quite useful, and will sit well in a mix.

The same holds true for digital synthesizers. You sit down in the local music store to try one out, and every factory patch sounds lush, rich, and full. And almost totally useless in a band setting.

However, if you take the time to learn how to build your own patches, you will probably find that the raw samples and waveforms are pretty usable, and some very good sounding patches can be built from scratch, which are extremely good, are more realistic, and will work well in a mix.

There is something else to consider in the synthesizer patches that claim to be realistic samples of real instruments – and that is that they almost always far less high end and upper midrange than in the real instrument. This adds to the difficulty of getting your instrument to fit in the mix. There are other limitations which, in my mind, are design flaws. One of these is the stubborn refusal by synthesizer makers to put effects in a logical place in the signal chain. A great example of this, and the one that is a huge limitation on Roland synthesizers such as the JV-1080 and XP-50, is Roland’s insistence that a Leslie is an “effect” rather than a “speaker.” As anyone who has ever played a Hammond Organ knows, a Leslie is a SPEAKER, and is therefore the last thing in the audio chain – AFTER the amplifier. The Roland Leslie simulator is not too bad, but by putting it so early in the chain, they’ve made it almost useless. It is such things that make building good patches a real challenge sometimes.

While you can get some patches that will, in a mix, sound “kind of like” another instrument, such as, say, a trumpet; there are some instruments which you will never get close to, no matter how much time you spend creating a patch. A great example of this would be, of course, guitar.

So, know the limitations of the technology and your equipment, take the time to build your own patches, and you will get some very usable patches. Don’t expect to fool anyone into thinking that they are listening to a real vintage synth, a real orchestra, or a real guitar. Just get sounds that will accomplish the same purpose, and that sound good in the mix.

After all, the mix is what’s important. Nobody cares what an instrument sounds like soloed, because nobody but you can solo it. Everyone else in the world will only hear it in the mix.


Audio Purists

March 20, 2009

There is a group of sound guys I call “audio purists.” These are people who eschew anything which colors the sound, such as EQ. To them, “purity” of sound is the ultimate goal, above all else.

All EQ colors the sound, not only by varying the frequency response, but also, as a side effect, affecting the phase angle at certain frequencies. This is most apparent when boosting frequencies, and, in extreme amounts causes “ringing,” or resonance at the boost frequency. While this is clearly heard at extreme levels of boost, it is present to some degree at more moderate amounts of boost. This is what purists object to.

Because of this, a “purist” will, when using the 31 band EQ to tune the system, only use it to cut frequencies. In fact, there are a few graphic EQs on the market that are “cut-only.” They do not boost at all. In theory, this should be a good idea. But, what of a situation, where the system sounds pretty good, but has one frequency area that is slightly lower in response to the rest of the spectrum? Most people would simply use the 1/3 octave to boost those few frequencies the 1 or 2 dB needed to smooth things out. The “purist,” however, would prefer to cut all other frequencies, to avoid boosting any band. However, one thing often overlooked is that cutting adjacent bands does NOT result in a flat response. For example, if you cut every band on a graphic EQ by 3dB, the resulting curve would not be a flat response which was simply 3dB lower than the input. Each band has the most affect at its center frequency, and gradual shoulders which boost or cut less and less the further from the center frequency you get. Also, phase shift problems are most apparent in these “shoulder” areas. These shoulder areas are additive, which means that the cut (or boost) of two adjacent bands combine where the shoulders of the filters for those two bands overlap. Therefore, the result from pulling every band down 3dB would be a response which was down 3dB at each center frequency, with ‘ripples’ between bands of lesser or more attenuation. In addition, there would be phase shifts across the entire spectrum. The end result is that, to avoid boosting in one small area of the frequency range, you would be introducing an odd frequency response in the entire spectrum which would be filled with phase anomalies. This in the name of “audio purity?”

A related thought is that they will refrain from using any (ANY!) channel EQ. In fact, they will switch it out of the circuit. This idea actually has some merit, of not carried to the extreme. Their thinking is that EQ is bad, for the reasons I stated above, and since they aren’t going to use the channel EQ, they might as well take it out of the signal path. Since every circuit adds some small amount of noise, you can avoid adding it by not having that circuit in the path at all. Consider that each EQ adds some noise, if you remove the EQs from all 32, 48, 64, or however many channels you are using, this can add up to quite a bit of noise you are avoiding. The thought goes that you should get your sound solely from mic choice and placement. In a situation where you want the most natural sounds possible, and if you are working with acoustic instruments where a “natural sound” is desired, this may be possible. I agree that you should do everything you can to get the sounds you need with mic choice and placement, but in a live situation, it’s not always possible, and you are most likely not working with cellos, violins, violas, etc. So, what is a “natural” sound for a synthesizer, electric guitar, or bass? Odds are, you are going to want to do some EQ to each one, or you will end up with a lifeless and unexciting mix. Can you imagine the average kick drum in a rock mix if you couldn’t have any EQ on it? Cheap EQ sections can sound pretty bad, but if you have good channel EQ, there’s no reason not to use it. Every board sounds different, and the EQ is a big part of this. It is one of the major things you should listen carefully to when shopping for a new mixer.

Another technique that is rather common, or at least was, among the purists,  is to put all of the channel faders at the +/-0 line, and do all of the mixing with the mic trim pots. I’m not real sure where this technique came from, other than the “purists” see a point on the fader where it is neither attenuating or boosting, and they figure that is the “natural” (or “neutral”) spot. However, if you read one of my earlier posts, you may remember that, on instruments that need to be quieter in the mix, this can result in added noise, since you are turning it down at the trim pot, and then running the fader at 0. Whatever noise is added by the channel’s electronics would be better reduced by getting a good strong signal through the channel, and then running the fader at -15dB, or where ever you need it. In my opinion, it is far better to get as hot of a signal as you can coming in to the channel. During sound check, have the player go through his loudest part, and use the PFL meter to set the input level to 0VU and do your mixing with the channel fader. That way, you have plenty of signal to work with, and if you are also sending monitor mixes from the FOH console, makes it MUCH easier to deal with.

A lot of the ideas that the “audio purists” have are based on situations in the mythical ‘ideal situation,’ but don’t often translate well to the real world of mixing a rock band. As always, use your ears and judge for yourself, but keep these things in mind when some “helpful” purist starts making suggestions. Try everything, and keep what works for you – just always consider every aspect and consequence of every technique you try. Otherwise, you may not know what is causing other, seemingly unrelated problems.


Finally Heard the Bose “Poles”

February 25, 2009

After reading the impossible and improbable claims in recent Bose ads about their new system that looks like a couple of black poles, I was, to say the least, skeptical (what? ME, skeptical??).

To be honest, I’ve never liked Bose stuff. Historically, most of it has consisted of a shitpot full of 5″ speakers in some sort of vented box, that, if given a few hundred watts, would get slightly above the level of a conversation while having no low end whatsoever, and no real high end either. They introduced some sort of pre-processing unit which I suppose was intended to smooth out the frequency response of the little 5″ speaker arrays, since they always had a horribly erratic response curve, with a peak in the 2KHz-5KHz area, which is not exactly pleasing to the ears. When you add one of the Bose processors to your system, it still sounds really bad, but now it also sounds really processed.

The Bose amps that they came out with years ago were grainy sounding and fragile. They apparently haven’t changed much, from what I heard.

To be fair, when I heard them, they were being used by a “DJ” who was playing MP3s, which don’t need much help sounding bad. But the overall sound quality was pretty dreadful.

Once again proving that “if it sounds too good to be true, it is.” Never believe marketing hype. Also, as I’ve said before, examine any spec sheets with care. And, above all else, LISTEN to something first if you think you may be interested in it. Preferably NOT in a music store. Find a band who uses the piece of gear you are interested in, and go give an extended listen. Also, talk to the soundman (NOT one of the band members). He can give you some insight into ease of setup and use, reliability, ease of repair, etc.


Sometimes I Just Don’t Get it

February 22, 2009

There are a lot of things I write in my blog that I expect to create controversy; to make people think. I know that there are some that won’t agree with some (or any) of what I say, and that’s fine. Most of the comments I get are positive, some say they don’t understand (which means either I didn’t do a very good job of writing my opinions, or maybe they just don’t want to understand). I’ve only gotten one nasty comment – and it wasn’t even about any of my political posts!

I got a comment which said “this post is bullshit” in response to something I had written, so I went to the post in question, and guess what it was:

How Loud Do You Want To Be?” was the target of this person’s anger. This post is about the most NON-controversial thing I’ve ever written, and yet someone felt strongly enough about it to take the time to express their displeasure.

For those of you who aren’t into pro-audio, or music, that post basically boils down to “make the music as loud as is appropriate for the type of music and the crowd” and gives some tips on how to deal with musicians (and drummers, too).

Oh well. I guess I can write that “politicians are a bunch of thieving liars and tyrants, who are only concerned with their own power, wealth, and importance” and that’s alright. But if I say “Don’t mix too loudly or quietly,” then it’s ‘bullshit.’

As I said: sometimes I just don’t get it.


What to buy?

February 19, 2009

One of the biggest questions you will run into when first putting everything together is what to get? There’s no easy answer, and it’s different for everyone because there are many factors that come into play.

How much money do you have? Your decisions will be a lot different if you only have $5000 than if you have $50,000. Not more difficult necessarily – just different.

What kind of music and venues do you want to work? An acoustic guitar duo in coffee shops is vastly different than a metal or prog band in 2000 seat clubs.

How big is your truck, and how much space are you going to have in anticipated venues? Are you going to fly any of it at any venue?

What kind of monitor system are you planning on providing? If you plan on providing only 3 or 4 monitors on 2 mixes, your requirements are going to be a lot different than if you want to be able to provide 6 monitor mixes plus sidefills.

If you’re going to be working with the same band exclusively, you can tailor your system to their needs, which makes it easier for you. But, if they aren’t playing regularly enough, or pay enough, to allow you to do this, you need to be able to meet whatever requirements you will run into as you work for a variety of artists, in a variety of rooms. If your capabilities are limited, you will lose work. Never, ever, mislead potential clients about your capabilities. It is better in the long run to not take a job than it is to misrepresent what you can handle.

Every part of your system can be a bottleneck which will limit what you can do, and how good it sounds while you’re doing it. Because of that, it doesn’t make sense to spend a huge amount of money on one part, and skimp on other parts. Like the proverbial chain, a system is only as strong as (and will sound as good as) its weakest link. It wouldn’t make sense, for example, to get a Heritage Series Midas console, and then only have enough money left to buy some cheapo speakers and amps. It would be far better to get a somewhat lesser mixer and spend more on getting adequate speakers and amps.

You’ll need to seriously think about what sort of system you want to be able to field. The big questions are: How many channels, how many monitor mixes, how many monitors. If well thought out, everything else is scalable. Bigger, or smaller, venues can be accommodated by taking in more, or fewer cabinets and amp racks, for example.

Don’t blow your mic budget on just a few expensive mics. It’s better to have 25 SM-58s in your mic kit than to have one U-87.

At this point, it might be good to mention that there is basically two general grades of equipment you will find from two entirely different markets. There is what is called the “MI” (Musical Instrument) grade gear. This is generally made by the same companies that make guitar amps, guitars, stomp boxes, etc. It also includes the “semi-pro” grade of equipment. These pieces are often built to provide features at the expense of sound quality or durability. It tends to be somewhat cheaper, and (more importantly) sound worse. It is not built as ruggedly as gear from the “Pro Audio” market. Pro Audio equipment is generally somewhat more expensive (even at the lower end of the pro audio scaled), is made by companies who specialize in pro audio gear (and may have been providing tour-grade gear for decades to the biggest sound companies), and tends to be much sturdier, and easier to repair.

You do NOT get “something for nothing.” You DO “get what you pay for.” You will have better luck (and better sound) with a bare bones piece of high quality equipment from a pro audio manufacturer than you will from an equivalently priced feature-laden offering from an MI grade piece. Why? I know that this sounds rather odd to say. The reason I say that is because the pro-audio piece is almost certainly going to sound better (often SIGNIFICANTLY better), is going to be much sturdier, and be easier to fix. You will be able to get longer useful life out of it because not only will it last longer, but the sound is going to be better, so you won’t be as anxious to replace it. In the long run, it will be cheaper.


Random Stuff About Speakers

January 1, 2009

Speakers are the last link in your audio signal chain, and can have the most dramatic effect on the overall sound of your system. They not only determine how efficient your system is overall, but also the frequency response, transient response, and how loud you can be.

A speaker consists of a very few parts.

1) The frame – simply the metal frame in which everything else is mounted. They can be stamped metal, which is the cheaper method, lighter, and potentially not as rigid; or they can be cast metal, which is usually used in the more expensive speakers.

2) The magnet – a magnet attached to the rear of the frame. The heaviest part of the speaker. The heavier the magnet is, the better the efficiency and low end response.

3) Voice coil – A coil of wire wrapped around a “bobbin” to form a cylindrical coil, which rides in a gap in the magnet. The two ends of the coil are connected to terminals on the frame of the speaker. The output of the amplifier is connected directly to the voice coil. When power is applied to the voice coil, the magnetic field in the gap causes the voice coil to move. (the stronger the magnetic field, the more movement for a given power input, all other things being equal)

4) The cone – The conical shaped thing that faces the front of the speaker. Traditionally, these are made from a type of cardboard. Sometimes you will see metal or plastic used as cone material, but not usually in pro-audio applications. The cone is attached to the voice coil, whose oscillations are transferred to the cone, which in turn, moves air.

5) The surround – This is the wavy part around the outside edge of the cone, just inside where the cone is glued to the frame. Is is most often cardboard, but in some speakers can be cloth, or foam rubber(!). The surround doesn’t really serve any audio purpose – it exists merely to connect the moving cone to the non-moving frame.

6) Dust cover – the little dome in the center of the cone. This keeps dust from getting into the voice coil gap. These can be vented, having a small (1/2″) section of screen in the center, to aid in cooling and to allow more unrestricted movement.

Barring physical damage, such as dropping the speaker, the most common parts of the speaker to fail are the voice coil, cone, and surround.

How they fail, and why

The surround can fail due to a couple of causes, neither of which is much of a mystery. The surround can just simply wear out. The cardboard of the surround is much thinner than the cone material, and, by necessity much more flexible. Over time, the constant flexing of the surround weakens it, and it can eventually just start developing rips radially. Or, in some closed-box applications, the air pressure inside the box can actually blow a hole in the surround. A hole in the surround doesn’t necessarily make a noise itself, but it can change the frequency response of a cabinet, since it is effectively a port. Also, once the rip or hole in the surround gets bad enough, the cone can become misaligned (off-centered) which will cause the voice coil to rub, which you will definitely hear.

A cone is more trouble-free, as long as you don’t stick a screwdriver through it while installing the speaker. Unlike the surround, a cone’s job is to remain rigid, so the entire surface area moves as one unit. Like the surround, a cone can, after time, wear out from the constant movement and humidity. The cone becomes flimsy and “soft.” Once this happens, the cone no longer moves as a single unit, which will affect the frequency response and efficiency.

Check your speakers every couple of months to make sure the cone and the surround are intact. If you are REALLY paranoid, rotate your speakers 90 degrees. Yes, if they hang the same direction over a long period, and are also exposed to travel bumps, the weight of the cone over a long period can possibly cause cone misalignment.

The dust covers aren’t usually a source of much trouble, although they can occasionally come loose because the glue holding them on dries out, and the combination of the dust cover’s inertia and the air pressure behind it can cause one edge to come loose. Do NOT try fixing them with tape (even duct tape isn’t going to work, believe it or not). Even if you can get it to stick, it’s not going to hold the dust cover firmly enough to keep it from rattling, so not only will the dust cover still rattle, but now the tape is going to rattle as well. If you have a rattle and find out that it is a dust cover, you can get through a show by CAREFULLY removing the dust cover entirely. Take it to your recone expert the next day.

Voice coils are the most common part to fail on a speaker, and there are two different results when they do: The speaker will rattle; or the speaker will be silent. (There is also the not entirely un-humorous “flaming speaker” which was usually caused by a Crown DC-300 dumping its DC supply into your speakers, but that’s another story)

All of the current flowing through the wire of the voice coil causes heat. This heat is mostly transferred to the metal of the magnet, which has a heat sink to dissipate the heat into the surrounding air. The movement of the speaker itself also helps keep it cool. However, if you exceed the power handling of the speaker continuously for an extended period, the voice coil will overheat. This overheating causes the adhesive used to bind the wire to the “bobbin” to come loose, the wire will partially unwind, and will either rub the voice coil gap, which you hear as a severe rattle, or will bind completely, which you hear as silence. There’s nothing you can do at the show venue. Replace it with a spare and take it to your recone expert when you can.

The voice coil can also jump out of its gap. If it remains aligned while it’s out there, it may slip back into the gap with no ill affects, other than you will hear distortion caused by the non-linear response of the speaker. If, however, it doesn’t quite remain aligned perfectly, it won’t hit the gap, and will instead slam down beside the gap, and will be essentially “locked up.” It will be silent. The cone won’t move when you push it. You MAY be able to pull the cone outward carefully (putting equal pressure on all sides of the cone) and get it to slip back into the gap, but don’t bet on it. Be aware, that since the speaker will not move, it will draw more current than normal, and since it doesn’t have the cooling action of the movement, will overheat more easily. In other words, if you notice a speaker not working, troubleshoot it as soon as possible.

As with all repair and maintenance work, if you have to open a cabinet, while you’re in there, check all electrical connections, and check the cabinet integrity. If you see any loose braces or stripped screws, FIX THEM. If they remain unfixed, they WILL be a future source of a rattle.