Quick Fixes For Better Sound

We all know the usual litany of ways to improve your sound: “Buy a better mic,” “buy a better preamp,” “buy a better . . .” Yeah, you get the idea. (My personal favorite is “write a better song,” but that’s beyond the scope of this article.) However, there are a lot of simple fixes you can do to improve your sound —

We all know the usual litany of ways to improve your sound: “Buy a better mic,” “buy a better preamp,” “buy a better . . .” Yeah, you get the idea. (My personal favorite is “write a better song,” but that’s beyond the scope of this article.)

However, there are a lot of simple fixes you can do to improve your sound — sometimes dramatically — that involve little or no money. And even if your ship has come in, these tips are still well worth following if you want to bring your recorded sound to a higher level.


Many people can hear some improvement with sample rates above 44.1kHz, but the question is whether the improvement is worth the extra storage space (and computer horsepower) — and if so, which sample rate to use. Few think that 176.4 or 192kHz is worth the effort not just because of issues like reduced track count, but also because it makes no apparent difference in sound quality.

While the high sample rate buzzword is 96kHz, sample rate converting down to the 44.1kHz of a Red Book CD requires some pretty fancy math — your 96kHz signal gets divided by 2.176870748299319727891156462585. Sure, in theory, today’s sample rate converters should be able to handle the math. But tests have shown there’s a great difference in generated artifacts among sample rate converters (for a fascinating white paper, go to www.bias-inc. com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf).

Instead of 96kHz, consider going with 88.2kHz (Figure 1). It provides virtually all the practical benefits of going with 96kHz, and downsampling to 44.1kHz simply requires your sample rate converter to divide by two. That may explain why some people think material recorded at 88.2kHz sounds better than material recorded at 96kHz by the time it ends up on a CD. In any event, give 88.2kHz a shot; if it sounds better to your ears, go for it.


We mean “24” as in bit resolution. If you’re still recording with 16 bits, flick that switch in your host now, and get with the program. Sure, your files will take up more space. But storage is getting less expensive these days, and computers are fast enough to process this extra data without giving too much of a hit to your track count and ability to use plug-ins. Twenty-four bits gives you more headroom while recording, better dynamic range, and more “footroom” as well. Besides, with most converters, to obtain an honest 16 bits of resolution you need at least 20-bit converters anyway.


Oxidized contacts can definitely affect sound quality in a couple of ways. One is that resistance can build up, which is equivalent to putting a resistor in series with your cable. A more insidious problem is when crystallization builds up, and those little diodes act like little crystal radios, ready to detect RF and inject it into your system as low-level hash.

The more patch points and mechanical switches you can get rid of, the better. For those that remain, use contact cleaner like Caig’s DeoxIT (www.caig.com) to keep your connections up to snuff. This can make a big difference, especially if you have a lot of patching in your studio.


For example, clicking on a check box within Sonar switches its audio engine over to 64-bit precision (Figure 2). If you’re piling on the tracks, doing complex mixes with a ton of automation, and want reverb tails to decay into nothingness, this can indeed make a difference. It doesn’t really stress out your computer, even if you’re using a 32-bit operating system. But with the world going to 64-bit computing from input to output, it makes sense to have your audio engine calculating in that world as well.


At the very least, resonances can be annoying. But even worse, you may mistake them as part of the sound coming out of your speakers. Bundle cables, caulk gaps, tighten down screws (and of course, turn off the snare on snare drums that aren’t in use!).


Most digital meters, while more accurate than their inertia-ridden analog counterparts, measure the instantaneous level of the samples that make up the signal — not the actual signal level that results from interpolating those samples. So, it’s entirely possible that the actual level is several dB higher than what the meter indicates, which means your signals could easily be going into clip-land occasionally without your knowing it.

Granted, some will say, “use your ears; if you don’t hear it, who cares?” But while you may not hear distortion on a single track, add together a bunch of mildly clipped tracks, and something may sound “wrong” — even if you can’t identify the exact cause of the problem.

So, give your peaks a little breathing room and treat –6 as max. Besides, with 24-bit resolution, you’re just throwing away an extra bit if you’re recording 6dB lower. This won’t make any significant difference in sound quality.


High clock speeds and dual core processors have pretty much put an end to the days of underpowered CPUs, but their legacy continues in many plug-ins that offer “high-quality” and “low-quality” options, with the latter placing less stress on your CPU. But you shelled out for that shiny new computer specifically to stress out your CPU, so seek out those “quality” switches (Figure 3) and turn them all up to the max quality possible.


We’ve become so accustomed to even-tempered tuning that most people have never even heard a truly pure interval without beating or tuning inaccuracies (other than an octave). So, if your synth does alternate tunings like just intonation, experiment (Figure 4). As long as you don’t have to modulate, you’ll hear stronger, more accurate chords and intervals. (Granted, there are ways to handle modulation so that this purity is maintained regardless of what you play, but few manufacturers see the value of implementing this — yet.)

As to inherently even-tempered instruments like guitar or piano playing along with properly intoned ones, don’t worry about it. The results will still sound better than having all even-tempered instruments.


This has been mentioned numerous times over the years, but just in case you missed it, use a sharp low-cut filter to roll off all unneeded bass frequencies: If the lowest fundamental in a track is 100Hz, start rolling off below that. It can really open up the sound of a mix.


They’re great for keeping spit from your lead singer out of the mic during live performance, but they affect the high frequency response and just plain don’t sound that good. If you don’t mic real closely, your mic has a low frequency rolloff switch, and your singer doesn’t get out of control, you may not even need a windscreen — try it. But if you do, get one of those round mesh deals. They cost more, but they’ll protect your mic while preserving its sound quality.


We’ll assume you’ve already placed your near-fields on stands, and made sure there aren’t reflective surfaces between the speakers and your ears (e.g., desktops, mixing consoles, etc.). But you still may have problems because of sound coupling from the speakers to the stands, which then causes other surfaces to vibrate. Although you can buy decoupling pads, the cheapest solution is to gather together some of those thick, neoprene promotional mouse pads you never use anyway and put them between the speakers and stands. If there was a lot of coupling going on, decoupling the speakers will result in a more focused, tighter sound.


Not all distortion is bad — just ask a guitarist. However, some digitally-generated distortion can have a harsh quality, even when it’s not supposed to (such as amp simulation software). One “magic bullet” I’ve found for smoothing out sounds like power chords is Adobe Audition’s Click and Pop Remover (Figure 5). Seriously. Depending on how heavily you apply it (I’d recommend starting with the “Hiss + Lots of Clicks” preset), it smoothes out all the spiky stuff.


This isn’t a low-cost fix so it kind of violates the article’s premise, but it’s worth mentioning anyway. I always knew that having properly conditioned power was good for your equipment, but never really believed it made an actual sonic difference until I reviewed the Equi=Tech balanced power system for EQ several years ago. Taking residual noise measurements with and without the Equi=Tech revealed about a few dB less noise with the Equi=Tech in use. It’s a relatively costly way to shave a couple dB, but every little bit helps — and good power filtering/conditioning helps promote happier, longer-lived gear anyway.


Not all EQ plug-ins sound the same. Run some signal sources through several different EQs at extreme, but identical, settings; for example, try boosting treble while processing crash cymbals, and determine which EQ gives the “sweetest” sound. Try boosting upper mids with vocals to find out which vocals get “harsher” and which ones simply get more present. Also experiment with cutting extreme amounts of mids to find out how various EQs hold up.


In a real acoustic space, reverb consists of millions of reflections. No matter how hard a reverb algorithm tries, it can only approximate that degree of complexity. Even convolution reverbs, while very realistic, cannot duplicate the sound of a real acoustic space — only simulate it.

One quick fix is to run two reverbs in parallel. For example, if you have a really good hall sound, run it in parallel with a plate sound (Figure 6). Each reverb will tend to “fill in the cracks” in the other one’s sound, producing a more complex and satisfying reverb effect. Also try combining reverbs in series; a lot depends on the types of reverbs you’re using. At some point while you’re experimenting, you’ll likely find a perfect combination of the two. Save both presets, because you’ll likely want to use them again.


It stands to reason that a $300 box isn’t going to include a $1,000 D/A converter on board, but many effects do include a digital out — and with quality conversion, you can hear what a device really sounds like.

Of course, if you have a suitable digital audio interface, you can feed the effect’s digital out directly to your computer. But sometimes, getting a little analog mojo into the signal chain — especially if it’s high-quality analog mojo — can add a character to the sound you won’t obtain by going digital-to-digital.


One common trick is to lower latency while recording, then kick it up to a higher value when mixing. This causes less stress on the CPU, thus allowing more plug-ins and virtual instruments to run, as well as providing the bandwidth to handle complex automation and other tasks.

However, some engineers swear that using more latency than you really need doesn’t help the sound, because it causes buffer timing issues that have the same kind of effect as using a loose clock signal: smearing of the sound, and narrowing of the soundstage. I don’t know of any hard proof about this, but there’s enough anecdotal evidence floating around that this concept deserves a closer look. Meanwhile, it’s probably a good idea to use no more buffering than is really needed, even if you’re mixing.


Different dithering algorithms are subtly different. But when you’re dealing with something that’s happening around the noise floor, it’s hard to quantify exactly what’s going on.

To judge how dithering affects the sound, record something acoustic with a long decay, like a decaying piano chord with the sustain pedal up. Next, cut just the end of each track (say, where the signal dips below –65dB or so), turn the volume way up while being very careful to make sure no high-level noises can get into the mix, then apply various types of dithering with different noise levels and noise shaping. Decide which one works best for you — assuming you actually need any, as with today’s high resolution recording and computing options, dithering may do nothing more than add a layer of noise you don’t really need.


Taken on a track-by-track basis, you may not hear any hiss in a project. But add together a bunch of tracks with low-level hiss, and it’s not so low-level any more.

Fortunately, today’s noise reduction algorithms do a superb job of minimizing noise while maintaining transparency of sound. The less noise they need to get rid of, the better the sound quality. So if you’re using noise reduction to reduce noise that sits around, say, –65dB or so, you can bring the noise down to –80dB with virtually no audible degradation.

The key to good noise reduction is to take a “fingerprint” of only the noise. Often you can find this at the head of a track, or during silences in the middle. Subtract this from your audio, and do this for all your tracks; you may be startled by the kind of clarity this imparts to the final mix — it’s like removing a scrim in a theater production.