But our brains also use other cues to determine distance. For example, human hearing excels at matching a sound with the echoes and reflections it causes, to localize its source. And we can apply this principle to add realism when creating the sound stage in a mix.

The Speed of Sound

Consider this picture, and the accompanying audio samples below.

In the scenario illustrated above, sound from the guitar reaches the listener almost immediately, whereas the reflections off the rear wall make a 40-foot round trip, and therefore arrive 40ms later. (Sound travels approximately 1 foot per millisecond.) With the drum kit, on the other hand, the direct and reflected sounds arrive at almost the same time.

The series of events goes something like this:

Our ears and brain are sensitive to these differences in sound arrival time, and use the information (along with other cues, like volume) to judge where a sound source is located in the space around us. Our brains know that sounds and reflections arriving together at our ears must have originated close to a wall, where sounds that arrive much before their reflections must be close to our ears.

Hear it in practice

Here are two short instrumental samples, both mixed from the same raw tracks, to illustrate how this can apply in a mix.

In the first sample, I’ve placed the drums closer by adding a delay between the direct drum sound and the reverb, so the reflections arrive 40ms later than the direct sound – which tricks our ears into hearing a 20ft distance between the drums and rear wall, as with the guitar in the above diagram:

In the second sample, I’ve simulated moving the drums further back by having the direct sound and reverb occur together, both 40ms later than the guitar.

Note that the levels are the same in each clip. I changed the delay times only, to illustrate the effect.

Issues

Caveat: The illustration above is grossly over-simplified. Sounds in a real room reflect off all the walls and surfaces, not just the rear wall. And our ears depend on much more than just timing differences to determine distance. But for the technique at hand, those complications generally aren’t important. The idea here is to trick listeners’ brains by exploiting a property of their sense of hearing, and whether there’s one wall or 4, human ears and brains interpret reverberant sounds the same. (If your listeners are mostly non-human, then all bets are off.)

Implementation: In Sonar, I configure sends (i.e. busses) with delay plugins for each delay time that I need, and I route tracks accordingly. But any platform that allows bussing or routing the signal can accomplish the same end result. So long as you can independently control the delay on the direct sound and on the reverb, you can manipulate the relationship between the two as described above.

Other levels: In practice, you’ll also reduce the level of the drum kit somewhat to make it sound more distant, and adjust the reverb level as required to make the effect more obvious.

Pre-delay

As an addendum: Most reverb units and plugins have a pre-delay setting for controlling the delay between the input sound and the reflections it generates. Pre-delay serves exactly the same function as placing a delay between the direct sound and the reverb. In essence, it “moves” the sound further from the simulated reflecting surface. So if your reverb unit or plugin supports pre-delay, you can accomplish much of the above technique without a separate delay plugin.

And remember this simple guideline when using reverbs for realistic 3d sound stages: To bring a sound forward in the mix, increase the pre-delay.

See Also: Reverb on kick drum, Reverb possibilities

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There are some great reasons to heed this advice. But they’re not immediately obvious, especially if you’re unfamiliar or uncomfortable with parametric EQs, and they’re rarely fully explained. I’ll explain and demonstrate below, and you can use the information to improve your EQ adjustments, and in turn your mixes.

Background

In brief, equalizers alter the tonal quality of audio by applying gain to a specific frequency range. (For something a little less brief, Sound On Sound’s Equalisers Explained is the best EQ primer I’ve read.)

Every EQ filter has 3 settings: Frequency, Gain, and Bandwidth.

Frequency determines where in the tonal spectrum an adjustment occurs. Low frequencies correspond to bass sounds, high frequencies to treble.

Gain determines the magnitude of the adjustment. Positive values increase the signal level at the specified frequency, and we call this a “boost.” Negative gain values decrease the signal level, and we call this a “cut.”

Bandwidth allows us to choose the range of neighbouring frequencies that our adjustment affects. Bandwidth is usually called “Q” (for esoteric reasons from filter theory.) Higher Q values affect fewer frequencies, and we refer to this as a “narrow” filter. Low Q values, on the other hand, yield “wide” filters that affect many frequencies.

This is easier to understand as a visual:

The diagram above shows 4 key combinations. From left to right:
#1 – A narrow cut – Note the high Q value, and negative gain.
#2 – A narrow boost – Note the positive gain.
#3 – A wide cut – Note the low Q value.
#4 – A wide boost.

Your EQ plugin may not look the same (for comparison here’s the above illustration using Reaper’s EQ) but all parametric equalizers support the same 3 basic options: Frequency, Q, and gain. And using these options, we can “cut narrow, and boost wide.”

But why is it good advice?

In practice, wide EQ cuts remove more signal, and therefore more of a sound’s defining characteristics. Remove too much signal, and the audio you’re treating no longer sounds like itself. This can certainly produce interesting effects, but it won’t yield accurate mixes.

Narrow surgical cuts, on the other hand, remove only specific frequencies, and as such leave the signal largely unchanged. The narrowest cuts can be practically inaudible, as they remove so little from the sound. Often, we use narrow cuts to remove only “problem frequencies,” such as ringing overtones from a drum or boomy resonance from an acoustic guitar, without affecting the overall character of the sound.

It might seem the same should be true of boosting – that narrow boosts are the least audible. But in fact, because of how our ears work, narrow EQ boosts usually sound unnatural and jarring, where wide boosts are much less obvious. (The reasons behind this involve science a little beyond the scope of this article. Summarized: Human brains evolved an innate understanding of the harmonic series, and narrow EQ boosts affect specific harmonics, producing timbres that we sense can’t possibly have occurred naturally.)

The effect should be clear in the examples below. These 5 audio files illustrate the various extreme EQ adjustments. First, an untreated track:

In the next sample, I’ve used a narrow boost at 2060Hz. [diagram]

The ringing is immediately apparent, and sounds unnatural and distracting. (Your ears and brain sense, based on the other frequencies, that there shouldn’t be a loud harmonic at that frequency.)

Now, here’s a wide boost at 2060Hz. [diagram]

While the sound might not be great, the ringing effect introduced above isn’t apparent, because the boost affects so many other frequencies:

The next example illustrates a wide cut at 2060Hz. [diagram]

Notice how much of the guitar’s character disappears:

Finally, in this example the narrow cut is barely audible at 2060Hz.[diagram]

All we’ve done is remove the ringing frequency, though since it wasn’t readily apparent in the original sample, its removal is hard to hear.

Caveats & Footnotes

These examples were contrived to illustrate an effect. (i.e. You’d never actually apply at 14dB boost at 2060Hz to an acoustic guitar track.) However, the principle applies regardless of the audio with which you’re working.

Note, too, that this technique is relevant only to adjustments made with parametric equalizers. Graphic EQs have a fixed bandwidth at each frequency, so “narrow” vs. “wide” cuts aren’t possible.

Finally, and perhaps most importantly, the advice is generally useful but NOT a set-in-stone rule. Sometimes, a ringing effect or hollowed-out sound is exactly what a mix requires. As with everything in audio engineering, let your ears be the final judge of what works best.

See Also: The Rule Of Mixing, General EQ Guidelines

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usually a “ring” isnt in one frequency…it’s a complex combination of frequencies. so you may need to eq out 2-3 different places. if you find a resonance, and eq it out, but still hear a ring, then repeat the process till all rings are gone. Then, give the snare a little boost in the mids to fatten it up, add a hint of distortion, send it to an aux with a plate verb…whatever it takes.

If you’ve never recorded outside your own home studio, you’ve likely found yourself wondering: How does the solo’d snare drum mic sound in a pro studio?

the mic picks up what you point it at. Your ears sit a good what.. 2-3 feet above the snare itself. When you hit a snare you hear a snare mic 3 feet below you and the rest of what you hear (a good portion) is how the room reflects the snare sound. When you have a mic an inch away from the snare itself..it’s picking up what your ears would hear an inch away from the same space. The snare sound you’re used to is compiled of the sounds from the top head, the bottom head, the walls, the ceilings…. and so on and so on

Tips for a fat, warm snare drum sound:

One thing that will help is a real consistent drummer. Grab some big sticks….. 2B Rock…. whatever. Tune the drum pretty loose with a little bit of muffling. If you’ve got a dynamic mic that is a little bottom heavy, try it out. I’ll use a ATM63HE, got the snap of a 57, with less honk and more balls. Light compression going in…. thats that….

And finally, some advice on reducing snare drum bleed on the kick mic …

The drummer is hitting the snare really hard and the kick really quiet. Tell him to stop. Moving the mic back will help because it puts the shell of the kick between the mic and the snare. You might also be using too much compression on the kick.

… and reducing other stuff bleed in the snare mic:

Is whoever that’s playing the snare hitting it properly? They should be attacking the shit outta the snare, make that hooker pop. On the flip side tell whoever it is to settle down on the cymbals, no need to mash them. It may be their thing and thats fine, just tell them to tone down their thing just a bit for the good of the recording. Playing properly will cure most of your micing problems.

Under specific conditions, which I describe below, the DAC can produce an analog signal that momentarily exceeds the level of the digital signal from which it was converted. This is known as an inter-sample peak, and while it may at first seem just a curious side effect of the conversion process, these peaks have implications for anyone working with digital audio. And in particular, engineers who like “hot” mixes.

Digital Audio – grossly over-simplified

Computers process numbers, so to work with audio in a computer, we must convert the sound into numbers. This is accomplished by “sampling” the audio signal at regular intervals – 44,100 times a second for CD audio – and saving the observed level. The diagram below illustrates this, albeit crudely. The samples in red, numbered 1 to 9, represent a sequence of signal levels observed by the recording interface.

You can see something similar by zooming-in on a waveform in any recording software. The “step-ladder” model illustrates the discrete samples of digital audio, with levels ranging from “-inf” (silence) to 0 dBFS (the loudest sound a digital system can represent [*]) However, sound outside a computer doesn’t move through the air in jagged, abrupt steps. Rather, it moves smoothly and continuously – more like the blue line in this diagram.

Digital to analog conversion, then, is essentially the creation of the smooth blue analog line from the set of stepped red digital samples.

Intersample peaks

Occasionally, the smoothing yields an interesting result, illustrated below. Note that in order to generate a smooth curve between samples 5 and 6, the DAC produces a signal that peaks higher than either of the samples. This is an inter-sample peak.

This diagram also illustrates the main issue with these peaks: Samples 5 and 6 are both at 0 dBFS – that is, they represent the loudest sound the digital system can reproduce. Yet the peak analog signal reconstructed by the DAC exceeds this level.

In short, the DAC in this example has generated an invalid signal.

How does it sound?

What this means in practice depends on a few factors, including the quality of the DAC, and the signal chain after the converter.

In the worst case, the example above would result in audible clipping when the system tries to generate the illegal voltage. But even if no clipping occurs, the analog side of the DAC will only handle the signal cleanly if the DAC’s analog circuitry has some headroom. If the DAC’s designers assumed that 0dBFS is the loudest signal the converter will emit (technically, a valid assumption,) then an analog peak above this level will cause distortion.

Here’s a short clip I deliberately mixed hot. There’s no digital clipping (the maximum sample level is -0.1dBFS,) but there are lots of inter-sample “errors”, including a nasty string of them at 0:13.

If you hear any pops or clicks, it’s a good sign that your setup doesn’t allow for inter-sample clipping. Most likely, though, the track sounds just fine to you. Many systems, especially those equipped for mixing audio, allow some head room between the maximum digital signal level and the onset of analog distortion.

But not every system… If you mix hot, especially if your meters peak within 1dB of full scale, your mixes probably contain these peaks. And though your DAC protects you, there’s no guarantee that your listeners have quality converters. In other words, you may be sharing clipped or distorted audio without realizing it!

Preventing inter-sample clipping

There are at least 3 ways to ensure your mixes don’t generate inter-sample distortion:

1. Leave lots of headroom in your mix: This is the obvious solution, and it requires no special processing. As long as you keep the peak levels in your mix below 0dBFS, the DAC will never encounter the situation described above. The consensus I’ve found is that real-world inter sample peaks never exceed 1dB, so keep your levels below -1.0 dBFS and you’ll be fine.

2. Use a mastering tool designed to prevent these peaks: iZotope’s Ozone is one such tool. The loudness maximizer has a one-click option to prevent inter-sample clipping.

3. Use a metering tool that highlights the peaks: SSL recently released the X-ISM Plug-In which adds an inter-sample peak meter to any VST-capable DAW.

X-ISM uses significant processing to provide a combination of up-sampling and filtering that mimics the operation of an oversampling DAC’s reconstruction process. The result is a meter that shows inter-sample errors and provides a useful tool that most DAW metering misses.

More reading

Along with the page on SSL’s site, above, here are a few more links that explore the issue in detail:

- 0 dBFS explained

- Chris Tham’s Issues with 0dBFS+ Levels On Digital Audio Playback Systems

- This classic thread on ProSoundWeb (especially the posts by Paul Frindle) presents head-spinning amounts of information. I recommend re-reading it every few months.

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The ideal drum reference tracks feature few other instruments, as musical instruments tend to mask frequencies in the snare and kick drums. But since drums aren’t often featured solo in pop and rock recordings, it can be tricky to find usable passages.

So these raw tracks of Dave Grohl playing on QotSA’s Songs for the Deaf should save some time! In fact, they might be the best commercial modern rock drum reference I’ve heard, as they sound like final mix stems, essentially the same drum mix used in the released track.

And for a classic sound, check out raw John Bonham drum tracks, outtakes from Zepplin’s In Through The Out Door sessions. Unlike Grohl’s samples above, these are compressed more than in the final album mixes, but they still make a great reference for tones and overall kit balance.

First, the effect of pickups on guitar tone:

Even though we each have different ideas about our ultimate tone, I think we’re all looking for a rich sound – rich in harmonics, that is. Lots of harmonic content = lots of ‘tone’. If you have lots of harmonic content to start with, you can easily use other sound shaping tools (tone controls on the amp, in particular) to sculpt your favourite and unique sound. It’s a bit like giving an artist every colour he could wish for to paint a picture. If you only give him a pencil, he can still draw a great picture if he’s really good, but has limited options.

The principles of rock guitar tone:

The best rock tone is from saturated power tubes directly driving a guitar speaker hard, with no load or attenuator getting in the way. The only really satisfactory way to get actual cranked tube amp and speaker tone with almost no room noise is to use a speaker isolation cabinet and its attendant gear.

Finally, the mother of all pages: Derek Miller’s collection of articles about guitar tone in rock’n'roll:

Perhaps the stereotype rock tone is that of the Marshall stack: a rectangular, 100-watt (or more), tube-powered amplifier “head” stacked on top of two speaker cabinets, each containing four 12-inch speakers. In this case, the guitar is a bit less important to the overall sound, although most who prefer it use Gibson-style solidbody guitars like the Les Paul or Gibson SG, with dual-coil “humbucking” pickups. Cranking up the Marshall creates a buzzing, distorted, complex, and extremely loud sound.

See Also: Mic placement and tone video

There’s a more important bitrate for most home recordists, however: The number of bits you use to record raw tracks. In all likelihood, your recording system gives you two choices, 16-bit and 24-bit. So which should you use?

Tweak has the most accessible discussion of the subject I’ve read. Short answer: Record everything you do at 24-bit. The article spells out a few good reasons, but here’s the meat of it:

You can record at lower levels, with more headroom. This ensures that the occasional peak is not truncated at the top and it will give converters some room the breathe. Because you are not pushing the limits of your bandwidth, your instruments will sound clearer, and the vocals may sound “cleaner”, the song will mix better and there will be less noise.

(The article also touches on the optimal sample rate, and while I agree with the conclusion, it’s for different reasons. For more details, see the discussion of the myth that higher sample rates yield more accurate recordings.)

A microphone’s ability to accurately capture these transients is known as transient response, and it’s an important property to consider when selecting a mic. To understand why, think of how a microphone works.

Diaphragm and Transient Response

All studio mics operate on the same basic principle: Sound energy moves a diaphragm, and the diaphragm’s motion is converted to an electrical signal which can be measured and recorded.

Diaphragms differ from mic to mic. Dynamic mics have a coil or ribbon, where condenser mics have a lighter capacitor membrane. (For background, see Wikipedia’s page on microphones and the comprehensive microphones section on GSU’s Hyperphysics site.) Regardless of type, however, the motion of all diaphragms is governed by the laws of physics. Specifically, inertia: Lighter diaphragms require less energy to move than heavier diaphragms. Consequently, lighter diaphragms react quicker than heavier diaphragms to abrupt changes in sound energy. That is, they have a faster transient response.

Generally, we find the lightest diaphragms in small diaphragm condensers (SDC’s) while large diaphragm dynamic (LDD) mics have larger moving-coil diaphragms. As such, SDC mics are more responsive to transients than LDD mics.

This diagram, taken from Shure’s indispensable Microphone Techniques for Music – Studio Recording, illustrates the response of a condenser mic and a dynamic mic to an electric spark impulse.

Though the difference is small, around 10 microseconds, the condenser mic (top line) responds more quickly to the impulse. Further, the dynamic diaphragm takes longer to stop moving after the impulse has passed. Note the continued “wobbling” on the right of the graph.

Diaphragm “stop” time

The difference in response is even more pronounced when viewed on a larger scale. To illustrate, I rigged an example with 3 microphones,

• Studio Projects C4 – an SDC
• Apex 205 – a cheap ribbon mic
• Shure SM58 – a small diaphragm dynamic mic

Here’s how the microphones respond to the click from a pair of drum sticks:

The differences are striking. The C4 (on top) has what I’d characterize as the “cleanest” response. The SM58 (on bottom) took about twice as long to “settle down” after the sound had passed.

And the ribbon mic (in the middle) has a completely different extended response. You can practically see the ribbon itself flapping back and forth inside the mic, taking almost 10 milliseconds to settle.

Choosing the right transient response

It’s important to note that just because a microphone has a faster transient response, it’s not necessarily a better mic. As always in recording and mixing, your ears are the final judge of “better,” and sometimes you’ll simply prefer the sound of a sluggish diaphragm. Many people, for example, opt to use ribbon mics as drum overheads precisely because the ribbon’s response softens harsh-sounding cymbals.

There are a few general guidelines, however, when considering how a microphone’s transient response will affect your recordings:

Compression: Larger diaphragms, with their slow response, tend to naturally compress a sound, smoothing out the transients
Smearing: Additionally, since large diaphragms take longer to stop moving after a sound has passed, they can also smear transients, sometimes blending one into the next.

These effects combine, in varying degrees depending on the mic, to yield a dark or flattened sound, generally suitable for bass, electric guitar, and edgy vocals.

Detail: Condenser mics, especially SDCs, better represent the transients we hear and, as such, yield a more detailed sound.
Higher frequencies: High frequency sounds tend to produce sharper transients, in which smaller diaphragm mics are better at capturing nuance.

These effects combine to yield a brighter, crisper sound, generally appropriate for acoustic guitar, drum cymbals, and delicate singers.

I’ve seen this all summarized as:

Faster – brighter, slower – darker.

Of course, that’s a very general guideline, because there are other important properties of a mic to consider (e.g. polar pattern, frequency response.) But when choosing a mic, it always helps to remember how its responsiveness will colour the sound of the instrument you’re recording.

First thing is to do the faders up listening. If you’re a member of the band, or the engineer, or even worse both (as well as the song writer and the overall aranger of the songs….like I am), then TRY REALLY HARD to forget that. You have to become the mixer and listen to the track with fresh ears. Once you done the faders up listen a few times, you will have a pretty good idea of how the track goes …

Here, the folks on Gearslutz discuss modest low-end microphones. They list some great options if you’re in the market for a decent pro-sumer microphone:

I’m a drummer and I’ve always been satisfied with a D112 on the kick and an SM57 on the snare. I’ve got a very old pair of Oktava mk012s that I love for overheads. I’ve also used these for room mics and acoustic guitar with nice results… SM57 is good enough honestly on a snare. If you can’t make a snare pop with a 57, it’s not the mics fault.

(The thread introduced me to Sweetwater’s customers top mic picks page, which, depending on how you view Sweetwater’s customers, could also be a handy reference.)

Finally, here’s a thread with some interesting thoughts on modulating tape speed to change a recording’s character:

if you were to take the original tracks and layer them with the slowed down tracks (providing that the tracks weren’t *that* slowed down and could occassionally be resynched) then yes, the sound will get thicker. Simply slowing the tape speed down though will do absolutely NOTHING to “fatten” the sound. No additional overtones are added, no distortion should be added (provided your tape machine is properly calibrated) and even if it were, I doubt it would be the euphonic type of distortion you’d be looking for …

I’m not sure how effective the trick would be for fattening the sound, especially in a tape-less DAW, but if you’ve got a “try anything once” mindset here’s Audio Mastermind’s list of free pitch-shift VST plugins.

Previously: The Big Page of Mix Tutorials

I was certain the space would yield a better drum sound. Still, I thought it would be interesting to hear how big a difference the room actually makes. So I took lots of measurements, and recreated the drum and mic configuration when I got home.

First, though, here’s a rough mix from one of the drum tracks I recorded at the cottage:

I used a standard arrangement: Recordman overheads; kick, snare, and floor tom close-miked; and my T4 as a room mic, in omni mode about 15 feet from the kit, up high. My home studio doesn’t have anything near 14-foot ceilings, but it’s spacious enough that I could get all the microphones, including the T4, the same distance from the kit that I had them in the cottage.

Here’s the same piece recorded after I got home:

Again, this is the same drum kit, tuned the same way, recorded through the same microphones, played by the same drummer. Literally the only thing that changed is the room in which the drums sit.

The difference, predictably, is most obvious in the solo’d room mic. Here’s how the drums sounded through the room mic in the cottage:

And here’s how they sound through the same microphone, at the same distance, in my home studio:

The clearest differences are the snare drum, which sounds much bigger in the larger room, and kick drum, which sounds like a different drum altogether in each recording.

The end result? The drums sound a little more natural in the high-ceiling, all-wood room. So the space matters, obviously.

I have a fairly large home studio, and it’s well acoustically treated, so I wasn’t expecting a night-and-day comparison – and this isn’t. But while either of these mixes would make a serviceable drum track, depending on the mix, I prefer the stuff I recorded in Muskoka, as it’s a bit more open and natural-sounding.

Perhaps the real lesson here, though, is that every recording decision affects the final product. Just as small tweaks can improve a mix, big changes, like traveling 2 hours north of the city with your drum kit and recording gear, also add up!

Previously: Better drum mixes with a drum reference track, Parallel compression for fatter drum tones

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