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.

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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.


Speakers (Part 1)

October 20, 2008

This is the last link in the chain of which you have control. It also happens to be probably the most complex issue when it comes to choice, placement, and a host of other issues.

When it comes to speaker selection, we’ll assume that you have listened to a variety of models, and have found several that sound good to you. We’ll also assume that you are going to buy existing full-range systems from any one of several makers. The first thing you need to consider is: how are they going to be used? What kind of rooms are you most commonly going to be in, and how large are the audiences going to be. A small (50 seat) club which is a wide room requires a different type of cabinet than a large, long venue, or an outdoor show.

You also need to take into consideration how loud you need it to be at various points in each venue.

In small rooms that are not very deep, you may be best served by front-loaded cabinets, which have a wide dispersion angle. This means you can use fewer cabinets on each side of the stage, while still having the majority of the audience in the “near field” (hearing sound directly from the speaker rather than merely the reverberant field). If you have found 2 or more systems that you like and can’t decide which to get, compare the efficiency ratings of each. Pick the one with higher efficiency – you will require a lot less amplifier power to achieve the same SPL (Sound Pressure Level, or “loudness”). For instance, one rated at 100dB is going to sound twice as loud as one rated at 97dB. If you pick the one rated at 97dB, you will have to run your system 3dB louder to compensate – which requires TWICE the power. Every 3dB increase in loudness requires a doubling of amplifier power! But beware – amplifier companies often don’t give you the full specs, especially on lower-priced amps, or amps from the musical instrument manufacturers. An efficiency rating should give you not only the SPL, but also at what input wattage (usually 1 watt), distance from the speaker at which it was measured (usually 1 meter), maximum distortion level at measured SPL, and over what frequency range (with +/- listed as well). The “standard” distance is now 1 meter – a calibrated SPL meter is placed 1 meter away from the speaker, on-axis to take the measurement. Next is the frequency range at which the measurement was taken. It is easy for the manufacturer to get deceitfully high efficiency ratings if they limit their measurement to only one frequency. This should also give you how much variation there is in this frequency range. The high and low frequency limits are typically the points at each end of the spectrum where the response is 3dB down (-3dB). There will be frequencies within this range that vary from a perfectly flat response, and that maximum should list listed as well (such as +1dB). So a full specification for a full-range speaker system should look like: 101dB(a) SPL, 1W/1M, +1/-3dB.

How does this translate to “how loud will it be 100ft away?” That also depends on a lot of things. It of course depends on how much power you put into it. In the above example, to get to 111dB at 1M, you will need to put in ten times the power, because an increase of 10dB requires ten times the power. So, 100W input will get you to 111dB at 1M. So, how does that translate over the distance? It’s hard to give a general rule because all rooms are different. If you used this cabinet outdoors, where there is no sound reflection, each doubling of distance results in a 4-fold decrease in loudness (6dB) (the “inverse-square” rule). So, at 2 meters, 111-6 equals 105dB. Double the distance again to 4 meters and you get 105-6= 99dB. At 8 meters, you end up with 99-6=93dB. At 16 meters (around 50 feet), you are left with 93-6=87dB – not very loud for your 100 watts of amplifier power. Also consider that you need to have headroom, or you will end up with clipping on every drum hit, or loud passage in the music. The more headroom you have, the more dynamic range you will have. To have a measly 10dB of headroom, you are going to require ten times the power if you are using that one speaker. 100W x 10 = 1000W. Whoa. There’s a problem. There are a couple of things to look at to fix this.

Look at a speaker system that has higher efficiency. If you go with one that’s 3dB more efficient, you can cut your power requirements in half. You can also look at a speaker system that concentrates more of the sound in the area you want. This is where dispersion angle comes into play. A front loaded cabinet has a very wide dispersion angle, and thus is good for a small, wide room, but doesn’t do a good job of concentrating the sound into a narrower angle. A horn loaded cabinet, on the other hand, is built to control dispersion, and as a side benefit, typically has a much higher efficiency. Ah ha! It sounds like this might be something to look into.

Everyone is familiar with the high frequency horn that has a 1″ or 2″ driver on the back of a flared “horn.” This horn directs the sound waves, and also increases the efficiency of how well the driver’s diaphragm movement is coupled to the air in the room. They will have a horizontal and vertical dispersion rating, a ‘cutoff’ frequency, and hopefully, a graph showing how well it controls dispersion at various frequencies (a common horizontal dispersion number is 90 degrees). As the frequency increases, the dispersion tends to narrow substantially. And, since the dispersion angle decreases, that means that the same amount of sound if focus on a smaller area, resulting in an imbalanced frequency response. This results in a situation where someone who is sitting slightly off-axis, but well within the 90 degree coverage angle, will hear less and less from the horn as the frequency increases. By the same token, people in the back of the room, who are only slightly off-axis (by a 10 or 15 degree angle), are going to have their heads cut off by the high frequencies.

(more later)


Using reverb live?

October 18, 2008

A lot of people have asked me whether or not to use reverb in a live situation. As always, my answer is to do what your ears tell you to, but here’s my opinion:

If your show is in a small (500-2000 seat) theater which was designed for plays, you may actually use a good reverb with excellent results. If, however, you are in a larger venue, usually the last thing you want to add is more reverb. In cases like this, I would be more inclined to use a very slight amount of a short delay. It sort of achieves the same result – that of adding some ‘space,’ but tends to not muddy things up quite as much. If you’re mixing an outdoor show, you can use reverb, but I find that it kind of sounds like it doesn’t belong. I suspect this is the psycho-acoustical aspect: your eyes see no reflective boundaries, and your ears don’t hear any reverb on any other sounds in the area, so a bunch of reverb on vocals or drums in your mix sound out of place.