WORKFLOW

Justin uses Pro Tools to record but after he’s cleaned up his tracks he’ll routinely bus out of the DAW and up the channels of the mixing console. In the analogue domain he has racks of outboard EQ, compression and other effects. Sometimes he’ll use Pro Tools effects and compression while at others he’ll use the outboard — often both.

“I find working entirely in the box laborious and time consuming,” reckons Justin. “I much prefer to move faders and get my hands involved. A hybrid approach is the best of both worlds.”

Part of the recent upgrade includes a new patchbay that provides ready access to all of Justin’s outboard gear, making the new console just that much more powerful at the mixing stage.

“I honestly think that patchbay is bigger than my imagination,” enthuses Justin. “To just have everything at my fingertips without needing to crawl behind the console or reconfigure things is a huge luxury.”

PREAMPS

The SSL Origin preamps are being fully utilised as well as the line inputs.

“I’ve always done that. I’ve always used the console pres to record through. Other studios will use various outboard preamps but, for me, especially recording drums, I like the sound of the same pre on every mic on the kit. I find it easier to mix.”

The sound of the Origin preamps have been a revelation: “They’re amazingly clear — you can hear a pin drop. The TAC Scorpion’s preamps were much darker, and introduced a lot of noise, I just don’t have any of that now.

“Then you’ve got the Origin’s Drive circuit, so you can get that bit of grit if you want it as well.

“The Origins channel EQ is great as well, I’m pretty sure everyone’s well aware of how powerful those EQs are.”

NEXT-GEN SOUND

Will the upgrades mean an opening up of the studio to a broader range of clients? Well, yes and no.

“We’ve never advertised,” comments Justin. “And because I’ve always had a ‘day job’ I’ve only ever wanted the studio to work on my terms. I enjoy mixing and recording bands that I’m into. I don’t enjoy being forced to sit through a day of listening to crappy music I hate. I’ll still be selective about what I do and I don’t do. I’m not doing it for the money.”

Some 20 years on, is Justin’s wife still happy with the move from the living room to the commercial studio?

“She’s okay with me spending so much time in the studio,” says Justin. “After all, it’s not like I’m down at the racetrack or the pub. My 15-year-old daughter is also catching the studio bug. She plays guitar and keys but she’s also showing an interest in audio engineering. I’m happy about that.”

THE AMM108

The AMM108 is the smaller model, almost cube shaped, and only a foot square in the old money. The 1.7-inch compression driver is mounted in front of an eight-inch woofer using a shared magnet design that helps maintain phase between the drivers, and reduces intermodulation distortion. The 110° x 60°  dispersion is shaped by the Bose Beamwidth Matching Waveguide, it looks like something out of ‘Predator’ but its purpose is to align the high and low beam widths through the (passive) crossover region. Weight is 13kg, power handling is quoted as 150W long-term/2400W peak at 94dB sensitivity for a calculated max SPL of 116dB (121dB peak).

The AMM108’s output belies its footprint. The first time I used it was as an in-fill for the front section of the audience and it had plenty of volume with outstanding clarity and intelligibility at all levels. Some EQ was needed, it had a chunky Lab-clone amp driving it and proved an ideal speaker for the in-fill application — small enough to preserve sight lines but also able to handle high level peaks without distortion, if required.

As a foldback wedge the AMM108 is instantly likeable. It looks somewhere between cute and tiny on an empty stage but when the band is set up and they’ve got pedals and other paraphernalia strewn across the stage, it looks just right — the other traditional wedges look unnecessarily bulky. Throw angle is the usual 35° and the outstanding coherence delivered by the coaxial design makes vocals super present with high resistance to feedback. The bands were impressed.

THE AMM112

The AMM112 is the larger full-range speaker and it’s a step up in every way. The 12-inch woofer and three-inch compression driver need a 50% larger cabinet than the AMM108 and it’s near twice the weight at 22kg. Power handling increases to 300W long term/2800W peak at 96dB sensitivity for max 121dB (131dB peak). The speaker can be switched to run passively through the internal crossover or bi-amped via an external crossover.

The first time I used the AMM112 was as a floor wedge with Ash Grunwald. Straight out of the box, a quick tune and, like the AMM108, the response was crisp and bright across the high-mids with no sense of being near the limit at average/high stage levels. The benefits of the coaxial design are evident working close to the speaker: traditional ‘horn on top of woofer’ designs exhibit noticeable phase changes between the components when the performers move around in front of the speaker, especially general purpose two-way speakers laid on their side. Coaxial speakers sound on-axis across the working area. Ash liked it, used it, and “it sounded awesome” was his comment after the show.

AMS115 SUB

Completing the range is the AMS115 compact sub. It’s a direct radiating, ported design with a 15-inch long-throw neodymium driver in an enclosure measuring 470 x 470 x 500mm and weighing in at a manageable 28kg. Frequency response is quoted at 40Hz–100Hz (-3dB) with power handling at 500W long term/2800W peak for 123dB/130dB SPL. It takes a bit more thought to get going. It seemed low in level compared to the full-range speakers at first but once pumped a bit it was a good match with impressive power handling and a fast, accurate response. An AMM115 sub with an AMM108 on top makes a great full-range system. I enjoyed using this combination as a delay system half way down the Castlemaine Town Hall for the 2022 Spring Ball.

CUT ABOVE

The AMM Series speakers can be used as portable systems but they’re beyond ‘throw-in-the-back-seat powered speakers’ type of portable. They’re the next step up in quality and price, and while they’re efficiently sized, they need at least amps and some processing. I’d suggest they’re better suited to professional production suppliers or installations where the AMM quality and scalability can be most valuable. Passive speakers often have advantages in these applications. Foldback is neater with passive speakers. Half a dozen of either of the AMM full-range speakers and one on top of an AMM115 sub for drumfill would make a powerful, flexible and coordinated foldback system (add more to scale up and throw in infill, if required).

Installations are made easy with a range of mounting options including 6 x M10 threaded inserts, a U-bracket and a suspension kit plus 4 x M8-threaded inserts for a pan-and-tilt bracket. The 110° x 60° dispersion can easily be changed to 60° x 110° by rotating the cabinet. For portable applications, both the AMM108 and AMM112 have pole mount sockets tucked away besides the handles. The Sub1 Adjustable Speaker Pole screws securely into the M20 threaded pole mount on the top of AMM115 sub. That’s a big improvement over the usual floppy pole sockets. The top half is like a regular speaker stand with adjustable heights and a locking pin, and that’s better than a fixed-length pole. 

Overall these are genuine high-power, multi-purpose speakers based on classic point source design in no-frills, practical cabinets. Bose Professional seems to be aiming for a larger space in the pro audio world, the AMM Series will give them some street cred.

WHAT ABOUT LEVELS?

All of the comb filtering frequency response examples shown so far were based on worst-case theoretical scenarios that assumed the original sound and the delayed sound were combined at the same magnitudes, resulting in complete cancellation at fc (the fundamental cancellation frequency). That’s rarely the case in practice. If the two signals are not combined with the same magnitudes then the cancellations at fc and above will not be complete nulls, and the reinforcements below and above fc won’t reach the full +6dB – therefore the comb filtering effect is less significant. The greater the difference between the magnitudes of the two signals, the less significant the comb filtering will be.

The illustration above shows what happens at fc when the original signal (green) is combined with a delayed version of itself (blue) that has a lower magnitude than the original signal – as would be the case if the delayed signal had travelled further than the original signal (e.g. comparing a close mic with a distant mic on the same sound source). In this case, the original signal’s positive half-cycles reach +1 at the same time the delayed signal’s negative half-cycles reach -0.5, and vice versa. The resulting signal (yellow) is simply the addition of the original and delayed signal. If both signals had the same magnitude then the result would be complete cancellation, but that’s not what we see above.

The illustration above is for the same situation as the previous example but the frequency is now 2fc (two times fc) so the delay time is now equal to one full cycle rather than one half cycle. In this case, the original signal’s positive half-cycles reach +1 at the same time the delayed signal’s positive half-cycles reach +0.5, and vice versa. The resulting signal (yellow) is simply the addition of the original and delayed signal. The combined signal still shows reinforcement, but it is not as strong as shown in the examples in the previous two instalments.

The illustration below shows what the comb filtering looks like in this example. At the frequencies that cause cancellation (i.e. fc and all odd-numbered integer multiples of fc), when the original signal is at the peak of its positive half-cycle the delayed signal is at the peak of its negative half-cycle. The original signal has an amplitude of +1, while the delayed signal has an amplitude of -0.5. The resulting signal’s amplitude is +0.5 (+1 + -0.5), which is half the amplitude of the original signal. When converted to decibels this represents a decrease of -6dB. In other words, the cancellations will reach a maximum of -6dB rather than being complete nulls.

What about the reinforcements? At frequencies that reinforce (i.e. below fc and at all even-numbered integer multiples of fc), when the original signal as at the peak of its positive half-cycle so too is the delayed signal. The original signal has an amplitude of +1, and the delayed signal has an amplitude of +0.5. The resulting signal’s amplitude is +1.5, which is 1.5x higher than the original signal. When converted to decibels this represents an increase of +3.52dB. Therefore the reinforcements will reach a maximum of +3.52dB, rather than the +6dB seen in earlier examples where both signals had the same magnitude.

As shown above, we still see the familiar comb filtering shape but the peaks do not reach +6dB and the dips do not form complete nulls. It’s still going to cause the familiar comb filtering problems, but they won’t be as severe or as obvious. As we reduce the magnitude of the delayed signal the affect of the comb filtering becomes less obvious, eventually reaching a point where it’s not an audible problem – although at that level the contribution of the delayed signal to the mix might be so small that it’s not worth using anyway.

PRACTICAL EXAMPLES & SOLUTIONS

Now that we’ve seen what causes comb filtering and how it manifests in our signals, let’s return to those three practical situations we looked at earlier – the guitar amplifiers and the drum kit – and see what the comb filtering is doing. Once we understand those situations we can apply the same understanding to other audio situations.

We’ll start with the example given previously that uses two microphones on one amplifier, because it’s the easiest to see how we get an original and a delayed version of the same signal.

Two Mics, One Amplifier

Here’s the illustration for this example from the previous instalment, but with different distances than used earlier.

This is a classic miking set-up for electric guitar amplifiers with quad box speaker systems. It uses a Shure SM57 up close to one speaker cone (0.1m, as in the previous example) to capture grunt and bite, and a Neumann U87 at a placement and distance of 1m to capture the ensemble effect of multiple speaker cones working together. Let’s do the calculations used in the previous instalment to determine fc, starting with the PLD (Path Length Difference) and following with the ATD (Arrival Time Difference). The PLD is 0.9m (i.e. 1m – 0.1m), and that gives us a corresponding ATD of 0.00261 seconds (i.e. 0.9 / 344).

We’ll stick to integer values for the answers to the following fc calculations to allow some wriggle room for the inevitable rounding errors that occur when working with decimal values of distance and time…

PLD: fc = 344 / (2 x 0.9) = 344/1.8 = 191Hz

ATD: fc = 1 / (2 x 0.00261) = 1/0.00522 = 191Hz

For this example fc is 191Hz, as confirmed by both calculations. It’s right between F^#₃/G^b₃ (185Hz) and G₃ (196Hz), and close enough to each to be a problem because the break-even frequencies will be 127Hz (120°) and 254Hz (240°) respectively – all notes that fall between 127Hz and 254Hz (i.e. from C₃ to B₃) will suffer cancellation in accordance with the comb filtering frequency response curve.

Cancellation will also occur at and close to all odd-numbered integer multiples of 191Hz (573Hz, 955Hz, 1.337kHz, etc.), and reinforcement will occur at all even-numbered integer multiples of 191Hz (382Hz, 764Hz, 1.14kHz, etc.), as shown below. Notes in the green areas will be reinforced, while notes in the red areas will be cancelled or reduced in level.

But what about the levels? If the signals from both microphones were mixed together at the same magnitudes, then the comb filtering would be as severe as shown in the illustration above. If they were mixed at different levels then the comb filtering won’t be as severe. For example, let’s say the Neumann U87’s signal was combined at 50% of the level of the Shure SM57. As we saw in an earlier example, in this situation the cancellations would only reach -6dB and the reinforcements will only reach +3.52dB. The resulting comb filtering response would look something like this:

Even when the delayed signal is mixed at 50% magnitude the comb filtering creates a significant change in the frequency response, meaning that the resulting signal will not sound like it did through either of the individual mics and, more importantly, it will not sound as the musician intended — the comb filtering is literally changing the levels of individual notes.

One solution to this problem, for recording at least, is to align the waveforms of the two signals in the DAW – otherwise known as time-alignment (as shown above). Slide the U87’s waveform along the time axis towards the left until it aligns with the SM57’s waveform. Zoom in to check that both signals start at the same time and that both signals cross the zero axis at the same time (to ensure there is no phase difference between them), and check that both signals have the same polarities at the same times as shown below.

The two signals should now add together with no comb filtering or polarity issues, allowing them to be blended together to create numerous tonalities and spatial possibilities. For example, the blend could favour the distant mic during rhythm/accompaniment parts to make the guitar feel a bit further back and behind other featured instruments, then changed to favour the close mic for moments that feature the guitar.

A more complex version of this time-alignment technique was used on the album Secret Samadhi (1997) for the band Live. A number of microphones were placed at different distances from the guitar amplifier, then each mic’s signal was timed-aligned (assumedly to the closest mic) and blended as desired to create a rich but singular guitar amplifier sound with many different tonal and spatial possibilities. [The song Lakini’s Juice provides a good example.]

Into The Concert Hall…

Let’s deviate from our three examples for a moment to see this microphone time-alignment technique used on a much larger scale…

Engineers who record classical, orchestral, chamber and choral music often use time-alignment with their microphones. In these situations the overall performance is usually captured by a main microphone array that is placed somewhere above and behind the conductor. It could be a stereo pair, or it could be a multiple microphone array intended to capture sound for an immersive/surround format. For this discussion we’ll keep it simple and stick to a main stereo pair. The placement above and behind the conductor puts the main stereo pair a number of metres away from the instruments, where it is intended to capture a balanced representation of the entire ensemble along with some of the room’s reverberation.

Instruments that are not captured loud enough or clear enough in the main stereo pair are given individual spot mics (i.e. close mics) that are mixed with the sound captured by the main stereo pair. The illustration below shows a spot mic being used to reinforce and/or clarify the sound of the flutes. To simplify this example we’ll use a coincident pair (e.g. XY, Blumlien, MS) as the main stereo pair, simply because it avoids ATDs between the mics in the main pair.

The considerable distance between the spot mic and the main stereo pair creates a large PLD and its associated ATD, and it is common practice to delay the spot mic to time-align it with the main stereo pair. This can be difficult to do because the waveform from the main stereo pair represents all of the music, and will therefore look very different to the spot mic’s waveform (which primarily represents the individual instrument it is aimed at). To solve this problem, prior to the performance a handclap or similar percussive sound is performed in front of each spot microphone and simultaneously recorded on the spot mic’s track and on the main stereo pair’s tracks. This provides a helpful visual cue for time-aligning the spot mic’s waveform with the main stereo pair’s waveform in the DAW, as shown below:

When widely spaced microphones are used as the main stereo pair (e.g. omnis spaced 50cm apart, aka ‘AB50’) and the spot microphone is significantly off-centre, as shown in the illustration below, the alignment handclap will not occur at precisely the same time in each channel of the main pair and it can be hard to decide which handclap to use for alignment purposes. Placing the spot mic’s handclap waveform in between the handclap waveforms shown in the main stereo pair’s tracks moves the spot-miked instrument to the centre of the stereo image and requires extreme panning to move it where it is supposed to be. It also introduces the possibility of two instances of comb filtering – one between the spot mic and the left mic of the main stereo pair, and one between the spot mic and the right mic of the main stereo pair.

This situation requires a more sophisticated solution. The spot mic signal is duplicated to create two identical tracks. One track is panned hard left and its handclap is aligned to the handclap in the left channel of the main pair, and the other track is panned hard right and its handclap is aligned to the handclap in the right channel of the main pair (as shown in the illustration below)

Keeping both spot mic tracks at the same level ensures that the spot mic’s signal appears in the correct place in the stereo image due to the time difference between the microphones, while preventing comb filtering with the individual mics in the main stereo pair.

The two versions of the spot mic signal might comb filter with each other when summed to mono, but in that case it means the two mics in the main stereo pair will also comb filter with each other when summed to mono because they’re the references that the spot mic signals were aligned to. The solution here is to correct the spacing of the main stereo pair, or, if it’s too late to do that, beat yourself up about it and remember to get it right next time.

Now, back to our previous three examples…

One Mic, One Amplifier

Let’s look at the first example from an earlier instalment of this series, reproduced below. It only uses one microphone, but a reflection off the floor creates a delayed version of the signal that combines with the direct signal at the microphone diaphragm – creating a comb filtering problem that we cannot fix with time-alignment because we don’t have access to the individual direct and reflected signals.

In this example a single microphone is placed 0.3m from the guitar amplifier but a reflection off the floor also reaches the microphone after travelling a total distance of 0.7m, creating a PLD of 0.4m – which corresponds to an ATD of 1.16ms. What is fc?

PLD: fc = 344/(2 x 0.4) = 344/(0.8) = 430Hz

ATD: fc = 1/(2 x 0.00116) = 1/0.00232 = 431Hz

Allowing for rounding errors due to the number of decimal places, both approaches give us essentially the same result: a fundamental cancellation frequency, fc, that we’ll call 430Hz. Cancellation will also occur at all odd-numbered integer multiples of 430Hz (1290Hz, 2150Hz, etc.), and reinforcement will occur at all even-numbered integer multiples of 430Hz (860Hz, 1720Hz, etc.).

[It’s worth noting that this problem can also occur in the previous ‘two mic’ guitar amplifier example, and cause considerable problems due to having potentially three different comb filtering situations: comb filtering due to the PLD between the microphones, comb filtering due to the floor reflection into the first microphone, and comb filtering due to the floor reflection into the second microphone.]

How bad will the comb filtering be in this example? The direct signal has traveled 0.3m to the microphone, while the reflected signal has traveled 0.7m to the microphone. According to the Inverse Square Law, the reflected signal will have 7.36dB less SPL than the direct signal. Here’s the maths for those interested…

ΔSPL = 20 x log(0.3/0.7) = -7.36dB

That’s equivalent to 43% of the magnitude of the direct signal. If the direct signal had a magnitude of 1, the reflected signal will have a magnitude of 0.43.

At the frequencies that cause cancellation (i.e. fc and all odd-numbered integer multiples of fc), when the direct signal is at the peak of its positive half-cycle the reflected signal is at the peak of its negative half-cycle. The direct signal has an amplitude of +1, while the reflected signal has an amplitude of -0.43. The resulting signal’s amplitude is +0.57 (+1 – -0.43), which is close to half the amplitude of the direct signal. When converted to decibels this represents a decrease of -4.88dB. In other words, the cancellations will reach a maximum of -4.88dB rather than being complete nulls.

What about the reinforcements? At frequencies that reinforce (i.e. all even-numbered integer multiples of fc), when the direct signal is at the peak of its positive half-cycle so too is the reflected signal. The direct signal has an amplitude of +1, and the reflected signal has an amplitude of +0.43. The resulting signal’s amplitude is +1.43, which is 1.43x higher than the original signal. When converted to decibels this represents an increase of +3.1dB. Therefore the reinforcements will reach a maximum of +3.1dB, rather than the +6dB seen in earlier examples where both signals have the same magnitude.

The resulting comb filtering response is shown in the illustration below. As with the earlier illustrations, musical notes in the green areas will be reinforced (in this case up to +3.1dB) and musical notes in the red areas will be reduced (in this case by as much as -4.88dB).

Because we do not have access to the two individual signals (direct and reflected), we cannot rely on time-alignment to fix this type of comb filtering problem. The incorrectly labelled ‘phase reverse’ (or similar nonsense) switch on our preamplifier won’t make any difference to the problem other than inverting the signal’s polarity.

The only solution here is to prevent it from happening in the first place. This can be done by placing absorptive material over the reflective surface – in this case, on the floor between the speaker enclosure and the microphone. The theoretical goal is to completely absorb the reflection, but it is often sufficient to absorb it enough to prevent it from causing noticeable comb filtering. Suitable absorptive materials include open cell foam (e.g. Sonex), compressed polyester fibre (e.g. Tontine Acoustisorb), crumpled blankets/duvets, cushions/pillows, and similar. To be effective the absorptive material typically needs to cover an area of at least 60cm x 60cm, centred over the reflection point.

Note that it is also possible for this problem to occur due to a reflection from a nearby wall or similar large reflective surface (this situation often happens when recording vocalists standing close to a wall, window or the reflective side of a studio baffle), and the same method of prevention applies.

Another way to prevent this problem is to change the position or placement of the sound source to increase the distance to nearby walls and other reflective surfaces. Also, if the speaker enclosure has a closed back it can be laid on its back, so the speakers face the ceiling, and miked from above (this might require the use of small blocks between the speaker box and floor to create enough clearance for the speaker connection). If placed in the middle of the room, the reflections from the walls and ceiling should arrive a considerable time after the direct sound and therefore have a considerably lower SPL – which is advantageous in minimising the audibility of comb filtering.

Alternatively, the speaker enclosure could be placed on a stand, raising the height and increasing the path length of the reflected sound, thereby reducing its SPL due to the longer distance and minimising the effect of the comb filtering. In the illustration below the speaker box has been raised by about 30cm to increase the path length, which lowers fc to 183Hz and gives the reflected energy 12.46dB less SPL than the direct sound.

This problem can also be solved by using a microphone with a bidirectional polar response, such as Royer’s R121 or BeyerDynamic’s M130 (shown below). Angling the microphone appropriately so that its side null is aimed at the reflection point will significantly reduce the amount of reflected signal captured by the microphone, thereby reducing the impact of the comb filtering. (We’ll be discussing how to use polar responses for minimising unwanted reflections in forthcoming instalments of this series.)

Combine the above polar response solution with absorption and/or raising the speaker enclosure on a stand, and the possibility of comb filtering drops remarkably.

Close-Miked Drum Kit

The third example given in an earlier instalment of this series was a close-miked drum kit, which typically results in three or more mics all placed within a metre or so from the snare drum’s mic.

The illustration above is from the example shown in an earlier instalment, but with distances and calculations added. Using the centre of the snare drum as a reference point, we can see that the snare drum’s microphone is 18cm (0.18m) from the reference point. Rack Tom 1’s microphone is 45cm (0.45m) from the reference point, creating a PLD of 0.27m (0.45 – 0.18 = 0.27) between it and the snare microphone. Due to the Inverse Square Law the snare’s SPL at this microphone will be 8dB lower than at the snare mic. When added to the drum mix so that Rack Tom 1 was the same perceived level as the snare in the mix, this mic’s signal will create comb filtering with the snare mic’s signal with an fc of 637Hz and a dip of -4.4dB. This will be accompanied by dips of -4.4dB at all odd-numbered integer multiples of 637Hz, and peaks of +2.9dB at all even-numbered integer multiples of 637Hz. Similarly we can see that Rack Tom 2’s mic will create comb filtering at 291Hz with dips of -2.3dB and peaks of +1.8dB, and the Floor Tom’s mic will create comb filtering at 358Hz with dips of -2.8dB and peaks of +2.1dB. (Note that the decibel levels for the peaks and dips given above are theoretical approximates only based on the notion that each mic’s level has been adjusted so that the individual drums are all approximately the same perceived level in the drum mix.)

When auditioned on its own the snare sounds focused and punchy, but when the other mics are introduced to the drum mix the snare becomes blurred and spongy – which should be no surprise considering that every other mic in the drum mix is applying its own set of comb filtering dips and peaks to the snare sound, as shown in the previous examples. Superimposing all of these peaks and dips over the snare mic signal’s frequency response makes a real mess of the snare sound when it is heard as part of the drum mix. The small time delays blur or ‘soften’ the attack transients, while the cancellations from the comb filtering create holes in the frequency response and the reinforcements on either side of the cancellations make those holes seem deeper – just like a sponge.

We’ve focused on the snare drum mic for this example, but it’s important to note that every mic shown in the illustration creates comb filtering with every other mic shown in the illustration, and each comb filter occurs at the fc that is determined by the PLD between the mics. This means that the comb filtering that occurs to the snare mic’s signal due to the contribution of Rack Tom 1’s mic may not have the same fc as the comb filtering that occurs to Rack Tom 1’s mic due to the contribution of the snare mic. It’s possible that the mix of the four drum mics in this example could contain up to 12 individual comb filters (three per mic), each with its own fc and its own sets of peaks and dips. We’re approaching the point where Lo-Fi becomes No-Fi…

Although not shown here, overhead mics can also introduce considerable comb filtering interactions with the close mics – especially if placed too low. Overheads can also create comb filtering between themselves if carelessly spaced apart, resulting in a phasey/swishing sound on cymbals that is easily identified when monitoring in mono.

Comb filtering is rarely a problem with a kick drum mic because a) it’s a sufficient distance from the other mics and b) it is often inside the kick drum and therefore isolated – except for spill from the snare bottom, which can be fixed with a low pass filter if problematic. If two kick mics are used on the same kick drum (typically one inside the shell facing where the beater hits the skin, and one just outside the shell) comb filtering between them can become a consideration, but this type of problem is easily solved with time-alignment as shown in the earlier example of two mics on the guitar amplifier because it is essentially the same problem: two mics at different distances from the sound source.

In contemporary popular music production the solution to the close-miked drum problems described above is to use noise gates when mixing live, and use automated mutes, editing and/or drum replacement when mixing in the studio. For other forms of music the solution is simple: use less mics on the drum kit. In situations where a more natural or accurate sound is required (e.g. orchestral, jazz and acoustic music) the entire drum kit is treated as a single instrument and captured with just one or two overhead mics, typically supported with a kick drum mic – not because a big thumping kick sound is required, but because the kick drum is considerably further from the overheads than the other drums and also aims outwards rather than upwards, so it is often poorly represented (at least tonally) relative to the drums and cymbals when captured by the overheads.

The illustration above shows a drum miking technique using a single overhead mic accompanied by a kick drum mic. A cardioid condenser is placed directly above the snare but angled to face somewhere between the floor tom and ride cymbal. Adjusting the angle of the microphone alters the balance across the drum kit, with the closer drums and cymbals arriving sufficiently off-axis to the mic’s cardioid polar response to be reduced by a dB or so — thereby balancing them against the more distant sounds from the floor tom and ride cymbal that arrive on-axis to the mic, while also maintaining a sufficiently detailed snare sound for brush work and similar. A ‘darker’ sounding SDC with good off-axis response, such as Milab’s VM44, works particularly well in this application. (We’ll be discussing how to use polar response for this kind of ‘mixing in the air’ in forthcoming instalments of this series.)

WHAT’S IT FOR?

Throughout this and the previous two instalments of this series we’ve explored comb filtering in depth, and learnt that the switch on our preamps that’s incorrectly labelled ‘Phase Invert’, ‘Phase Reverse’ or similar nonsense is not the solution for phase-related issues because it does nothing to the signal’s phase – all it does is invert the signal’s polarity. So why is it there and when should we use it?

To minimise induced interference, professional audio equipment uses balanced differential lines for analogue audio interconnections. These typically rely on three conductors: one carries the correct polarity version of the signal (indicated with ‘+’), one carries an inverted polarity version of the signal (indicated with ‘-’), and one carries a 0 volt reference (aka ‘signal ground’) and/or provides a shield against induced interference by enclosing the other conductors inside a flexible tube made of wire mesh or foil and referred to, of course, as the shield.

[It’s worth noting that in the early days of the XLR connector the intention was to use a four-conductor cable: one conductor carried the correct polarity signal, one carried the inverted polarity signal, one carried the device’s 0 volt reference, and the other provided the shield that enveloped the other conductors and shunted any induced interference directly to electrical earth before it had a chance to enter the signal path. This allowed the device’s 0 volt reference to be kept separate from the shield and electrical earth, which was a good way of minimising the possibility of ground loops. The 3-pin XLR connector that has become the industry standard still has four separate connections — three pins plus a solder tab to extend the cable’s shield to the connector’s shell — and some cable manufacturers still make the ‘three conductors plus shield’ microphone cable required for this type of connection, such as Belden’s 9398. However, contemporary designs use a three-conductor cable — two inner conductors and a shield that also carries the circuit’s 0 volt reference — thereby introducing the potential (pun intended) for ground loops. This has become known historically as the ‘Pin One Problem’. Google it…]

For many years before and after the introduction of the XLR connector there was no international standard detailing which pin carried the correct polarity signal and which pin carried the inverted polarity signal. Using the 3-pin XLR connector as an example, pin one was globally accepted as the shield and/or 0 volt signal reference but there was no standard defining which of the remaining two pins carried the correct polarity signal and which carried the inverted polarity signal. Some manufacturers and audio facilities settled on pin two for the correct polarity signal and pin three for the inverted polarity signal, and some settled on pin three for the correct polarity signal and pin two for the inverted polarity signal. This difference in wiring standards extended to microphone manufacturers, increasing the likelihood of polarity cancellation when microphones from different manufacturers were combined together. This wasn’t a problem for professional audio facilities where it was standard procedure for in-house technicians to check each new item of equipment and (if necessary) re-wire it to conform to their chosen standard, but it was a problem when equipment from different facilities was being combined – such as borrowing or hiring additional microphones or outboard equipment for a recording session or live performance. It was customary for audio facilities and services to have a number of incorrectly-named ‘Phase Reverse Leads’ (typically colour-coded yellow, and about 20cm long) or XLR barrels that connected pin two on the female XLR to pin three on the male XLR, and pin three on the female XLR to pin two on the male XLR. These short cables and barrels temporarily solved the polarity problem without altering the internal wiring of the equipment – which would not be acceptable with hired equipment.

The incorrectly-labelled ‘Phase Reverse’ (or similar nonsense) switch on the preamplifier solved this problem by swapping around the correct polarity signal and the inverted polarity signal at the point where the signal from the microphone enters the preamplifier, as shown below.

The confusion between pin two and pin three on the XLR connector was finally settled in the early 1990s when the AES14 Standard locked-in the use of pin two for the correct polarity signal and pin three for the inverted polarity signal. Nowadays all professional audio manufacturers follow this standard. However, that doesn’t mean we no longer need that incorrectly-named ‘Phase Invert’ switch…

WHEN TO USE IT?

Before going any further, let’s give this seemingly obsolete yet remarkably useful switch a more appropriate name. We’ll call it the Polarity switch. Why? Because it affects the signal’s polarity. There’s no need for any further clarification in the name because a signal’s polarity can only have one of two states: correct or inverted. As long as we know which switch position is ‘inverted’ we’re good to go. So when should we use it?

The most obvious example is in the event of an incorrectly wired microphone lead, patch lead or patch bay. If it’s a balanced differential connection and it is wired incorrectly between pins two and three, or between the tip and the ring on a TRS jack or socket, the received signal will be inverted polarity. That’s exactly the sort of problem the Polarity switch was intended to fix.

Likewise if we are using two microphones with one on either side of a drum skin, or one in front and one behind a speaker box. In both of these cases, the sound captured by one of the mics will be in the opposited polarity to the sound captured by the other mic. More about that situation in a moment…

A less obvious situation is when you are combining two versions of the same sound, one from a pick-up and one from a microphone. For example, combining the sound from an acoustic guitar’s pickup with the sound from a mic in front of it. The wiring of the pick-up and the use of a DI box in the signal path both introduce the possibility for a polarity difference between the pick-up’s signal and the microphone’s signal.

The same situation exists when combining a DI signal from an electric bass guitar with the signal from a microphone placed in front of the bass guitar’s amplifier, and also when taking a direct output from an instrument amplifier and combining it with a microphone placed in front of the amplifier or the instrument itself. In all of these situations it is worth changing the polarity of one of the two signals and alternating between both versions to see which provides the best sounding result. But wait, there’s more…

Combining the signal from a pick-up (via a DI box or amplifier direct output) with the signal from a microphone placed in front of the instrument or amplifier also introduces the possibility of a time delay between the two signals, because the distance between the microphone and instrument or speaker will invariably be greater than the distance between the instrument’s strings and the pick-up. This type of problem is easily solved in recording situations by using the time-alignment methods described earlier in this instalment. In sound reinforcement situations we have to rely on polarity inversion and perhaps a sample delay if using a digital console. More about that below…

ABOVE, BELOW, IN FRONT, BEHIND…

Let’s finish this exploration of polarity, phase and comb filtering with some common miking practices that create a combination of polarity and comb filtering problems, and require the use of polarity inversion and time-alignment to do properly.

Front & Back Amp Miking

Putting mics in front and behind a speaker enclosure that has an open back, such as a Roland Jazz Chorus combo amplifier, is an open invitation for polarity and comb filtering problems. Why? Because the sound emanating from the rear of the speakers is in the opposite polarity to the sound emanating from the front of the speakers. Also, the mic at the rear will almost certainly be a different distance from the speakers than the mic at the front, which means there will be an ATD (Arrival Time Difference) between the front and rear mic signals. When the mic signals are mixed together there will be cancellation due to the polarity difference between them, and comb filtering due to the ATD between them. The solution requires a combination of polarity inversion and time-alignment, as shown in the illustration below.

We must invert the polarity of the rear mic’s signal to put it into the same polarity as the front mic’s signal, and then we must time-align it with the front mic’s signal.

It’s a relatively simple process in the recording studio. For sound reinforcement applications the polarity is the most important part to get right. The time delay can be taken of by careful placement of the rear mic, ensuring it is the same distance from the speakers as the front mic to minimise the possibility of an ATD.

Now let’s look at something that’s not so simple…

Snare Top & Snare Bottom

In many genres of popular music it’s common to use two mics on the snare drum. One mic is aimed at the top skin (aka the batter head) and intended to capture the overall sound of the drum – including its fundamental note and harmonics/overtones, the player’s articulation (i.e. attack of the stick on the skin), rimshots, side sticks and brushes. This ‘snare top’ microphone placement usually captures enough of the sound of the snares vibrating against the bottom skin (aka the resonant head) to represent the entire sound of the snare drum – especially when using shells of 11.4cm (4.5 inches) or less.

When using shells deeper than about 17.8cm (7 inches) it can be difficult to capture enough of the sound of the snares with just a snare top mic. The deeper shell gives more body and fullness to the tuned part of the snare sound (i.e. the note from the top skin), thereby reinforcing its level in the snare top mic. At the same time, however, the deeper shell moves the snares further away from the snare top mic (compared to a shallower shell) which makes them marginally softer and duller. The result is a more subdued contribution from the snares and an overall ‘darker’ snare drum sound. If that’s not the desired sound for the music it can lead to excessive high frequency EQ boosting in an attempt to create a brighter and snappier snare sound – which has the unwanted side-effects of exaggerating spill from the hi-hat and cymbals, making rimshots and side sticks sound ‘clacky’, and bringing a ‘harsh’ and ‘scratchy’ tonality to brushes. To avoid these EQ-related problems a mic is placed underneath the snare drum specifically to capture the sound of the snares. Not surprisingly, it’s referred to as the ‘snare bottom’ mic.

The intention is to combine the snare top mic and snare bottom mic signals to create a brighter and snappier snare drum sound without requiring excessive boosting of the high frequencies and all of the associated problems described above. Although we like to think of each mic as capturing only its intended sound, the reality is that each mic will capture considerable levels of sound from both skins (which is, of course, why we often don’t need to use a snare bottom mic).

The illustration above shows a typical snare top and snare bottom miking scenario. The sound of the top skin goes directly into the snare top mic (green arrow), but also spills into the snare bottom mic (dotted green arrow). The spill into the snare bottom mic is an inverted polarity version of the sound going into the snare top mic and, due to the longer distance it travels, it arrives at the snare bottom mic a short time after it arrives at the snare top mic and thereby creates an ATD between the mics. Likewise, the sound of the bottom skin and snares goes directly into the snare bottom mic (blue arrow) but an inverted polarity version of it goes into the snare top microphone (dotted blue arrow) along with its own ATD.

When the top and bottom mics are mixed together we end up with cancellation due to the polarity differences between the two mic signals, and multiple instances of comb filtering due to the ATDs. The solution requires polarity inversion and time-alignment as shown earlier for the combo amp example, but, unlike the combo amp example, in this case we have two sound sources (top skin and bottom skin) creating two different instances of comb filtering, and that means we have to make an aesthetic choice between two different time-alignment solutions – we either align the spill in the snare bottom mic with the note of the top skin, or we align the spill in the snare top mic’s signal with the sound of the snares. This is discussed in detail at the end of this instalment (scroll down to ‘Making Great + Great = Great’).

CONCLUSION

In this and the previous two instalments we’ve taken a deep dive into polarity, phase and comb filtering as they apply to microphone use and placement. We’ve discussed the very important difference between polarity and phase, explored theoretical and practical examples of polarity and phase problems, seen the causes of comb filtering and how to fix them, and learnt why the terms ‘Phase Reverse’ and ‘Phase Invert’ (as commonly seen on mic preamps and channel strips) are nonsense. Whether or not you’ve understood the mathematics used in the examples is not important; what really matters is that you’re aware of polarity and phase problems, the differences between them, how to anticipate and/or identify them, and how to solve or prevent them.

Polarity is always a problem when miking something from both sides – such as top and bottom of a drum skin, or front and back of an instrument speaker. It is also a potential problem to be aware when we are combining a pick-up sound (via a DI box or similar) with a miked version of the same instrument or its instrument amplifier.

Comb filtering is a potential problem whenever we are combining a signal with a delayed version of itself. The range of human hearing is said to be 20Hz to 20kHz, so any comb filter with an fc between 20Hz and 20kHz will land within the range of human hearing and is therefore a potential problem. An fc of 20Hz requires a delay of 25ms (0.025s), and delay times of 25ms or longer are generally perceived as spatial effects (e.g. ambience, echos) rather than tonality problems. Similarly, an fc of 20kHz requires a delay of 25us (0.000025s); a delay time of 25us or less is going to cause comb filtering well above the capabilities of human hearing and can be considered irrelevant – unless we are capturing sounds with ultrasonic content and pitch-shifting them down to create sound effects. Delay times that fall in the range between 25ms and 25us create fcs between 20Hz and 20kHz, and we should always be aware of the possibility of comb filtering in those situations. Delay times longer than 25ms or shorter than 25us are probably not worth worrying about in terms of comb filtering…

Polarity is an amplitude-based parameter with only two states: correct and inverted. The Polarity switch offers an amplitude-based solution for polarity problems. Phase is a time-based parameter with infinitely variable positions, and time-alignment and/or microphone placement are the time-based solutions for phase problems. To reiterate something from an earlier instalment: using an amplitude-based solution (the Polarity switch) to fix a time-based problem (comb filtering) is like using SCUBA gear to go skydiving – you’re diving, but you’re using the wrong tools.

So remember: use the Polarity switch for polarity problems, use time-alignment and/or microphone placement to solve phase and comb filtering problems, and use both solutions when you’ve got both problems at the same time (e.g. snare top/bottom, amplifier front/back, etc.).

In the next instalment we’re going to take a short break from the maths and physics and look at polar responses. It’s only going to be a short break, however, because the inescapable reality is that audio engineering is grounded in maths and physics. If we want to understand how to use polar responses properly and effectively then we’re going to need to get, er, ‘physics-al’…

Review: Graeme Hague

Also of note is the Pro EQ3, which adds dynamic control to the EQ bands. Simply put, you can adjust the EQ bands as per usual, but with a threshold setting on each band to control when the EQ will engage. It’s a bit like a multiband compressor, but in reverse — and it’s going to be everyone’s go-to EQ over a master buss.

I’m sure we’ve all tried to wrangle a daunting de-esser that was difficult to operate, and that we weren’t able to achieve satisfactory results with. Thankfully, there’s a new De-Esser plug-in too. Studio One’s De-Esser is simple to use and gives impressive results — the default settings alone will get you most of the way there.

FLIPPING OUT

Another notable improvement in this release of Studio One 6 is the Fader Flip function in the Mixer section. Think of live digital mixing consoles and their ability to switch through layers on a desk. Similarly, with Studio One, users are able to switch or ‘flip’ the channel faders to control the send levels of any bus or FX channel. This allows for more precise settings, with the longer throw of the faders able to provide smaller incremental values. I also like to use the feature as a fault-finding tool — taking advantage of the fact that Fader Flip provides clear, visual reference to what signal is being sent where. So, if you’re ever having a ‘where the hell is that noise coming from?’ moment, try switching to Fader Flip and it will likely reveal the culprit.

A Customise option now allows users to pare menus down to the essentials. The new Smart Template takes the idea of pre-defining a song further by allowing users to build default layouts, channel configurations, VST instruments included at load-up, you name it.

Studio One comes in several varieties. Studio One Professional is the top-level version and can be bought with a perpetual license. The added functions make it worth the money, with the Project Page being particularly useful, as it provides everything needed to bring a collection of work together to create consistent digital releases. Mind you, the entry-level Studio One Artist is a surprisingly well-featured option for a cost-effective version, and the free Studio One Prime is no slouch — although the inherent limitations of a freebie version may soon become apparent.

SUBSCRIBING TO CHANGE

This brings us back to PreSonus Sphere, the monthly/yearly subscription version of Studio One. To lure users to the dark-side of the software subscription model, Presonus has included extra plug-ins, and gives users access to extensive loop libraries, cloud storage, and collaborative tools. Of these plugins, the Vintage Compressor, and the EQ emulations of the Fat Channel are my highlights. Another benefit of Sphere’s operation is that the PreSonus server doesn’t need a constant link, so an internet connection is only needed once every 30 days to keep things ship-shape.

PreSonus has worked to create a real sense of community around Sphere, to ensure that it’s more than just a subscription service. With Sphere, users can share ideas and have discussions with other members, work collaboratively on songs, and gain access to expert chat lines and video tutorials.

COMPLETE PICTURE?

So, does Studio One 6 give you absolutely everything you need? With the long list of included VST instruments and plug-ins, and the integration of Melodyne pitch-correction, it’s hard to think of anything that it lacks. The core workflow and GUI of Studio One is quick to get acquainted with, and a lot can be achieved before needing to learn about more complex features. The Studio One community is large and helpful, and PreSonus has produced a huge collection of video tutorials and masterclasses, meaning that help is never far away. By my reckoning, Studio One has become a complete DAW package, with the new Video Track and Lyrics Track ticking the final boxes.

If this is your first foray into DAWs — or even Studio One — then Sphere is a no-brainer. Subscription software is completely stable and reliable these days, and Sphere gives you full access to it all at a minimum of cost. Coming in at US$14.95 per month (US$164.95 per year) it’s on a par with those premium streaming services that we don’t think twice about paying for. There is also a 30-day free trial of Studio One Professional that is ‘coming soon’, if anyone needs to dip their toe in first.

Studio One’s future looks bright, devoid of doom and gloom — and no icebergs.