Welcome to No Limit Sound Productions

Company Founded
2005
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Our services include Sound Engineering, Audio Post-Production, System Upgrades and Equipment Consulting.
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Our mission is to provide excellent quality and service to our customers. We do customized service.

Tuesday, August 15, 2017

Q. Why do I sometimes see singers using two microphones on stage?

By Hugh Robjohns, Mike Senior & Martin Walker

I've noticed that, when I watch old concert footage, the singer is often using two mics. I'd always assumed that one was being used for recording purposes and one was being fed to the PA. However, I recently heard that the Grateful Dead used a two‑mic technique for noise cancelling, is this true, and how would it work?

Using two mics for noise cancellation. Combining outputs of both at equal gains, but in opposite polarities, will cancel ambient noise from each to a certain degree. 
Using two mics for noise cancellation. Combining outputs of both at equal gains, but in opposite polarities, will cancel ambient noise from each to a certain degree.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: You are quite right in that, back in the 1970s, it was quite common when recording a live concert for the recording people to simply tape a second mic to the PA vocal mic to acquire their recording feed. Mic splitters either weren't trusted back then or they were too expensive!

The two‑mic noise‑cancelling idea is well known and very common in many sound applications — most notably aircraft communication headsets — but is rarely seen these days in live sound and PA. The same basic physics explains why football commentators' lip ribbon mics work so well at rejecting the crowd noise.

The basic idea of the two‑mic technique is to have two microphones spaced a short distance apart (usually between one and two‑inches or two to six centimetres) in front of the mouth, or whatever the sound source is. Both microphones must be able to hear the sound source directly. If they are cardioids, they both need to face the sound source, although this is more usually done with omnidirectional microphones, for the following reasons. The ambient noise, being inherently diffuse sound, will be captured equally in level by both mics; their spacing will make no significant difference to the ambient sound level they capture. By contrast, the wanted sound will be in the near field of both mics and, provided the front mic is very close to the sound source (ie. near the lips of the vocalist), the inverse square law of sound‑energy dispersion means that the more distant mic will receive significantly less energy from the close sound source than the front mic will.

Accordions produce sound from two places — on most models from 'treble' and 'bass' ends — so it's important to recognise this when miking and set two microphones either side and slightly forward of the instrument. 
Accordions produce sound from two places — on most models from 'treble' and 'bass' ends — so it's important to recognise this when miking and set two microphones either side and slightly forward of the instrument.

This also helps to explain why omni mics are preferred in this role, because otherwise the close mic would have a far stronger bass response, due to the proximity effect, than the more distant mic, and odd tonal effects could result. By combining the outputs of both mics at equal gains but in opposite polarities, the similar level of ambient noise from each will cancel to a very large degree, whereas the significantly different levels of the wanted close sound from each mic will hardly cancel at all. Of course, there will be a slight level reduction of wanted sound in comparison to using just the close mic on its own but, given the 30dB‑plus of ambient noise reduction gained by this technique, that's usually a side‑effect well worth suffering when working in very noisy conditions.

The physical spacing between the two mics inherently introduces a small, but finite, time delay, and so when the two mic signals are mixed together, the frequency response will inevitably become comb filtered. However, if the distance between mics is only an inch or so, the first deep comb‑filter notch will be well above any significant, important component of the human voice, and the rest won't have any material effect on the sound quality either. To return to your original statement, this noise‑cancelling technique really requires two identical mics spaced a precise distance apart. Most of those old festival concert photos show completely dissimilar mics mounted with their capsules more or less coincident, which lends weight to the suggestion that they were for separate recording and PA feeds, rather than exotic noise‑cancelling techniques.

The Grateful Dead developed a version of this noise‑cancelling technique because of the very unconventional PA arrangements they used to employ, with all of the PA set up on stage behind the band as a 'wall of sound'. In this way the band heard exactly what the audience heard (no need for separate monitors!). It was a clever system, with each musical source having its own set of amps and speakers to improve headroom and minimise distortion.


Published July 2010

Saturday, August 12, 2017

Q. How do the different amp classes work?

I'm trying to learn a little more about amp design. One thing that really baffles me is the different classes available. What does an amp's class mean, and how does this affect the way it is used?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: In a Class‑A circuit, the active device (whether valve or solid‑state) passes current regardless of the polarity of the input signal; in other words, in an audio application, it is 'biased' so as to pass both the positive cycle and the negative cycles of an audio signal. The side effect of the biasing is that the active device has to pass current all the time, making it relatively inefficient.

In a Class-B circuit, the active device only passes current for one polarity of input signal — which polarity depends on the circuit design — and this makes it a much more efficient way of working. So, in this case, where it is required to pass a symmetrical audio signal using a Class‑B circuit, the circuit will need two active devices, one to handle each polarity. This is an arrangement often also known as 'push‑pull'.

Class C is a format that only conducts on signal peaks and is rarely (but occasionally) used for audio in situations where power efficiency is more important than distortion. Class D — which is now becoming very popular in audio applications — works by generating a stream of high-voltage pulses at a very high frequency. These pulses are modulated in such a way that the average energy they convey follows the wanted audio waveform.

Returning to the Class-B design, this exhibits a problem called crossover distortion for audio applications, because both of the active devices in the push‑pull pair turn off as the signal nears the zero line. The solution is to bias the devices so that they don't turn off. They actually continue to pass signal as it crosses over into the opposite polarity. In other words, it works a little more like a Class-A device (but without the same levels of power inefficiency).

 
In a push‑pull amp design, each active device handles one polarity of the input signal.

Hence the compromise name Class AB; it is a Class-B design biased to operate in a similar way to Class A around the crossover region. However, it should also be remembered that push‑pull designs can also be operated fully as Class A if required, and some high‑power amps do work in that way. This is also a handy technique for cancelling out even-harmonic distortion products in tube-amp designs.


Published August 2010

Thursday, August 10, 2017

Q. How can I improve acoustics in a long, thin room?

The diagram to the right shows my room, which serves as my studio. The dimensions seem to be bad for low frequencies and there are sound-pressure failures at 55Hz and between 110 and 140 Hz. I have an Auralex foam bass trap, but I don't known if absorption is the answer. What should I do to improve this situation?

 
If you have any choice of rooms for your studio, try to avoid those whose dimensions are multiples of each other.

Via SOS web site

SOS columnist Martin Walker replies: I agree: that's a bad choice for a room, dimensionally, as far as acoustics are concerned. The 2.6‑metre width and 2.5‑metre height are nearly identical, while the 5.8‑metre length is close to double these, giving you a shape that's almost two cubes joined together. The room is also relatively small, which will mean it'll have relatively few modes below a few hundred hertz and, as the dimensions are closely related to each other, these modes will pile up at some frequencies (resulting in a huge peak), with large gaps between them (creating big dips in the frequency response).

Room-mode frequencies are fairly easy to calculate, but it's even easier to plug your three dimensions into a utility, such as the on‑line MCSquared Room Mode Calculator (www.mcsquared.com/metricmodes.htm) or the Hunecke Room Eigenmodes Calculator (www.hunecke.de/en/calculators/room‑eigenmodes.html). However, if you've got a PC, the ModeCalc utility from Realtraps (www.realtraps.com/modecalc.htm) is one of the easiest to use, displaying the first 16 axial modes for each room dimension up to 500Hz in an easy-to-interpret graphics plot. It would show that the biggest gaps in your room mode plot occur between 30 and 60 Hz (which explains your hole at 55Hz), between 70 and 90 Hz, and again between 90 and 130 Hz (the other area you've already pinpointed).

Without acoustic treatment, your listening position will be very critical, since you can end up sitting in a bass trough at one frequency and a huge peak at another. However, your loudspeakers and listening position do look to be near their optimum locations for the flattest compromise response. The oft‑quoted ideal is to place your listening position (ears!) close to 38 percent into the room from the front wall.

Acoustic-foam bass traps, like the one you already have, can certainly be effective, and acoustic foam is also excellent for dealing with mid-/high-frequency early reflections from your side walls and ceiling. However, acoustic foam is invariably a lot less dense than the 60kg/m3 Rockwool that is generally recommended for DIY bass traps, and you will, therefore, require a much greater volume of it to achieve a similar amount of absorption at lower frequencies. In a small room, you'll simply run out of space before you can cram in enough acoustic foam traps to adequately deal with the problems.

Diffusion can be a good way to 'break up' your reflections so they become less troublesome, but you ideally need to be at least a couple of metres away from them to avoid hearing a set of discrete reflections, rather than a more diffuse soundfield, so they are not often used in small rooms like yours. Tuned traps also have their place in the grand scheme of things but, in my experience, tend to be more difficult to tune and place optimally compared with broadband trapping that you simply fit where the bass levels are loudest, so they absorb the sound more efficiently.

Overall, I think broadband absorption is your best bet; as much of it as you can reasonably fit into your room. Start by placing traps that straddle the front vertical corners of the room, then the rear vertical corners, followed by any other corners you can manage, such as the ceiling/wall corners, and even the floor/wall corners where feasible. Also, don't forget some side panels and ceiling 'cloud' at the 'mirror points' to deal with early reflections.


Published August 2010

Friday, August 4, 2017

Q. Is phasing affecting the sound of my double-tracked vocals?

By Various

I've been reading about how you have to be quite precise in matching the distance from source to mic when multi‑miking guitar cabinets, and something occurred to me. If this kind of phase alignment is so important in this instance, how can we avoid such issues when double‑tracking a vocal, given that the singer inevitably moves their head around? The singer in question here is me, and I tend to move around a fair bit when singing! I've noticed when lining up and trimming my doubled vocals in the past (and on my current song) that some words sound 'different' when combined than others, and by different I mean 'worse'. Could phasing be the underlying cause, and if so, is there anything I can do to rectify this?

 
Sorting your sample library into nested folders is an excellent way to help you find what you're looking for more quickly, but some software samplers (like NI's Kontakt 4, shown here) already provide extensive database 'tagging' systems for just that purpose.

Via SOS web site

SOS contributor Mike Senior replies: Yes, if you double‑track very closely, you'll inevitably get some phase‑cancellation between the two layers, but that's not a problem; it's an inherent part of what makes double‑tracking sound the way it does. However, the potential for phase cancellation between the parts won't be nearly on the same scale as with the two signals of a multi‑miked guitar amp, because, firstly, the waveforms of two different vocal performances will never match anywhere near as closely; and, secondly, the phase relationship between the performances will change from moment to moment, especially if you're moving around while singing. Furthermore, in practice a vocal double‑track often works best when it's lower in level than the lead, in which case any phase‑cancellation artifacts will be much less pronounced.

For these reasons, nasty tonal changes from double‑tracking haven't ever really presented a major problem for me, and if they're regularly causing you problems, I suspect you might be trying to match the layers too closely at the editing stage. Try leaving a little more leeway for the timing and see if that helps for a start — just make sure that the double‑track doesn't aniticipate the lead if you don't want it to draw undue attention to itself. Similarly, try to keep pitch‑correction as minimal as you can (especially anything that flattens out the shorter‑term pitch variations), because that will also tend to match the exact frequency of the two different waveforms. In fact, if there are any notes that sound really phasey to you, you might even consider shifting one of the voices a few cents out of tune to see if that helps. Anything you can do to make the double‑track sound less similar to the lead can also help, whether that means using a different singer (think Lennon and McCartney), a different mic, or a different EQ setting. You may only need the high frequencies to provide the double‑tracking effect, and these are unlikely to phase as badly as the low frequencies.


Published February 2010


Wednesday, August 2, 2017

Q. Do balanced connections prevent ground loops?

By Various

I've carefully wired up my gear using all balanced inputs and outputs, and proper balanced cables, but I'm still getting occasional digital hash in the background. What have I missed?

 
Even with balanced cables you can sometimes experience ground loops, so here's the best place to break one without risking RF interference.

Jamie, via email

SOS columnist Martin Walker replies: Ground‑loop problems can be absolutely infuriating, and I wrote a step‑by‑step guide to tracking them down back in SOS July 2005 (/sos/jul05/articles/qa0705_1.htm). In essence, you have to temporarily unplug all the cables between your power amp and mixer. If the noises go away, you've found the location of your problem. If not, plug them back in and try unplugging whatever gear is plugged into the mixer — and so on down the chain.

The majority of ground‑loop problems occur with unbalanced connections, so my next advice would have been to replace the offending unbalanced cable with a balanced or pseudo‑balanced version. However, as you've found, sometimes such problems occur even in fully balanced setups where you carefully connect balanced outputs of one device to balanced inputs of another via 'two‑core plus screen' balanced cables.
I recently had just such a problem in my own studio and, to make it even worse, it was an intermittent one, so whenever I got close to discovering its cause, it mysteriously vanished again. Here's what I did to track it down, so others can try some similar detective work in their own setups.

First of all, you've got to be systematic, and note down everything you try, particularly with an intermittent problem, so you don't have to start from scratch every time it occurs. In my case, I could hear the digital low‑level hash through my loudspeakers even with my power‑amp level controls turned fully down, and it also persisted when I turned off the D‑A converter box feeding my power amp. However, it completely disappeared as soon as I disconnected both cables between the D‑A output and power amp input.

These quick tests confirmed that the noise wasn't coming from the output of the converter, or from the power amp itself, but instead from a ground loop completed when the two were connected. However, just like you, I was already using balanced cables. I double‑checked the wiring of both of my XLR balanced cables and there were no errors: the screen of the cable was connected to pin 1 at each end, the red core connected to pin 2 at each end, and the blue (or black) core to pin 3 at each end. So far, so good.

Next, I double‑checked with a multimeter that there was no electrical connection between the metalwork of the two devices via my equipment rack (a common source of ground‑loop problems, and curable by bolting one of the devices to the rack using insulated washers or 'Humfrees'). Again, there was no problem.
The best wiring for balanced audio equipment is to tie the cable screen to the metal chassis (right where it enters the chassis) at both ends of the cable, which guarantees the best possible protection from RFI (Radio Frequency Interference). However, this assumes that the interconnected equipment is internally grounded properly, and this is where things can go awry. The cure is to disconnect one end of the cable screen, and the best choice to minimise the possibility of RFI is the input end (as shown in the diagram).

By this time, my intermittent problem had disappeared again, so here's another tip. I carefully cut the screen wire of one of my two cables just before it arrived at pin 1 of the XLR plug, but left the other cable unmodified. Then, the next time the ground loop problem occurred a few days later I quickly unplugged the unmodified cable, whereupon the noise disappeared immediately. This proved that I'd correctly tracked down the problem, and modifying the other cable in the same way ensured that it never happened again.


Published January 2010

Korg Education LP380 Digital Piano Lab Bundle

Tuesday, August 1, 2017

Q. How do I know a mic is worth the money?

By Various
What differences can you hear when comparing inexpensive and expensive equipment? As I do a lot of vocal recording, I'd like to splash out on a really good microphone. But how can I be sure that an expensive microphone is worth the money? What am I listening for?

Sarah Betts, via email

 
Fidelity and accuracy are expensive qualities to build into a microphone, so those are the areas that will generally improve as you increase your budget. However, this doesn't necessarily mean that your voice will sound better through a more expensive mic; it's more important that you find the right mic to suit you. Bono, for example, famously favours the inexpensive Shure SM58 over high‑end alternatives.
 

SOS Technical Editor Hugh Robjohns replies: The benefits extend far wider than just the sound, but basically you're listening for an improvement over your current mic, and you then need to decide if the price justifies that improvement, bearing in mind the law of diminishing returns. Going from a very low‑budget mic to a mid‑range mic will usually bring about very obvious sound improvements. Going from there to a high-end model will bring smaller improvements, which may not always be obvious. And going from there to a mic worth several thousand dollars will bring smaller benefits still. Some people will believe the improvements are worth the expense, others won't!

However, you'll know immediately and quite instinctively when you find a mic that is well suited to your voice, and that doesn't always mean the mic needs to be expensive. If you're looking for a general-purpose mic, expensive usually equates to increased flexibility in use. But if it's a mic that will always be used on your voice and nothing else, finding a mic that suits your voice is the prime directive.

Sonic fidelity or accuracy is generally an expensive thing to engineer into a microphone, and the most expensive mics are generally pretty accurate. But recording vocals is rarely about accuracy. It's more to do with flattery, and different voices need to be flattered in different ways. When working with a new vocalist, I'll usually try a range of mics to see which one works best with their voice. Sometimes the most expensive mic gives the best results, but it's equally likely that it will be a less expensive model. U2's Bono famously records his vocals using a Shure SM58, and he seems happy with the results!

But, as I said, there's more to an expensive mic that just the sound. More expensive mics tend to be built to higher standards. They tend to include internal shock-mounting for the capsule, to reduce handling noise. They are thoroughly tested to comply with the design specifications and provide consistent results. Being better constructed, they tend to have longer working lives and can be maintained by the manufacturer relatively easily. They also generally deliver a very usable (although that might not necessarily equate to 'the best') sound whatever the source, without needing much EQ to cut through in the mix.

Less expensive mics often sound great on some things but terrible on others, often needing a lot of EQ to extract a reasonable sound within a mix. Often they're less well manufactured, which reduces their working life expectancy and, once broken, can rarely be repaired.


Published March 2010

Monday, July 31, 2017

Q. What exactly is ‘headroom’ and why is it important?

By Various
I'm a synth guy getting more and more into recording and mixing my own tunes. One thing that stumps me is the issue of 'headroom': for example, in the case of my Focusrite Saffire Pro 26 I/O, the manual says that using the PSU rather than Firewire bus power yields 6dB of additional headroom in the preamps. I assume that this is a good thing, but how so? What is headroom and why do I want more of it? How do I know it's there (or not there), and how can I take advantage of it?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: These are all good questions. Every audio‑passing system (analogue or digital) has two limits: at the quiet end there is the noise floor, normally a constant background hiss into which signals can be faded until they become inaudible; and at the loud end there is clipping, the point where the system can no longer accommodate an increase in signal level and gross distortion results. The latter is generally due to the signal level approaching the power supply voltage levels in analogue systems, or the coding format running out of numbers to count more quantising levels in digital systems.
Obviously, we need to keep the signal level somewhere between these two extremes to maximise quality: somewhere well above the noise floor but comfortably below the clipping point. In analogue systems, this is made practical and simple by defining a nominal working level and encouraging people to stick to that by scaling the meters in a suitable way. For example, VU meters are scaled so that 0VU usually equates to +4dBu. The clipping point in professional analogue gear is typically around +24dBu, so around 20dB higher than the nominal level indicated on the VU meter.

That 20dB of available (but ideally unused) dynamic‑range space is called the headroom, or is referred to as the headroom margin. It provides a buffer zone to accommodate unexpected transients or loud sounds without risking clipping. It's worth noting that no analogue metering system displays much of the headroom margin. Rather, it's an 'unseen' safety region that is easy to overlook and take for granted. In most digital systems, the metering tends to show the entire headroom margin, because the meter is scaled downards from the clipping point at 0dBFS. The top 20dB or so of a digital scale is showing the headroom margin that is typically invisible on the meters of analogue systems. As a result, many people feel they are 'under‑recording' on digital systems if they don't peak their signals well up the scale, when in fact they are actually over‑recording and at far greater risk of transient distortion.

The reason why your interface offers greater headroom when operating from its external power supply is because the PSU provides a higher‑voltage power rail than is possible when the unit is running from the USB power supply. A higher supply voltage means that a large signal voltage can be accommodated; in this case, twice as large, hence the 6dB greater headroom margin. More headroom means you have to worry less about transient peaks causing clipping distortion, and generally translates to a more open and natural sound, so it's a good thing.


Published February 2010

Korg Education SP170S Digital Piano Lab Bundle

Saturday, July 29, 2017

Q. What’s the best way to add a subtle vinyl effect?

By Various
I'm trying to figure out how I would create a really old‑style, warm‑sounding distortion/crackle on a string motif for an intro to a song I'm writing. I'll be using East West Quantum Leap Symphonic Orchestra for the actual string loop, and I want to create a sort of 'AM radio' feel for it. That's easy enough to achieve using various EQ techniques, but I also want to give it a really subtle '60s record‑player crackle — something that's there if you know what you're listening for, but not so 'in your face' as to sound cheesy or clichéd. I was wondering if there are plug‑ins that can do this. I fear I may have to break the bank again...
 
Here are three plug‑ins you could use to add simulated vinyl noise to your audio tracks without breaking the bank: Izotope's Vinyl (left), Retro Sampling's Vinyl Dreams (far left), and Steinberg Cubase's bundled Grungelizer (top).

Via SOS web site

SOS contributor Mike Senior replies: There's no need to break the bank for this, because there are actually a few different freeware plug‑ins that provide the kind of thing you're after. One of the best known is Izotope's freeware Vinyl plug‑in, which is available for both Mac and PC. The advantage of this one is that you get a lot of control over the exact character of the vinyl noise you're creating: not only can you balance various different mechanical and electrical noises, but you can also choose the decade you want your virtual vinyl to hail from and how your processed audio is affected by disc wear.

The downside of this plug‑in for me, though, is that it doesn't seem to output some of its added noises in stereo, irrespective of how I set up the controls, and a lot of the character of vinyl noise, to me, lies in its stereo width. To be fair, though, the 'dust' and 'crackle' components seem to be stereo, and stereo was, of course, only really in its infancy in the '60s, so this might not matter to you. Indeed, collapsing the whole signal to mono might be a useful way to 'date' the string sound itself. If you're running Steinberg's Cubase, the built‑in Grungelizer plug‑in provides a similar paradigm to the Izotope plug‑in, albeit with a simpler control set. However, all the added noises from this plug‑in appear to be in mono too.

For stereo vinyl noise, check out the freeware plug‑ins from Retro Sampling (www.retrosampling.se). Both Audio Impurities and Vinyl Dreams can overlay vinyl noise, although you only get wet/dry knobs, so you're stuck with the preset effect. That said, if you set up the plug‑ins on a separate channel in your sequencer, you can dramatically adjust their character with EQ to make them seem less obtrusive — a combination of high‑cut and low‑cut filtering usually works well for me. If you want a smoother vinyl noise (less of the Rice Crispies!), you can also slot in a fast limiter or dedicated transient processor to steamroller spikes in the waveform.

These processing techniques also allow you to get good mileage from the vinyl noise samples that periodically crop up on sample libraries. I've been collecting vinyl noise samples for a while, so I can tell you that there are good selections on the Tekniks Ghetto Grooves and Mixtape Toolkit titles, as well as on Spectrasonics' original Retrofunk collection. I've also turned up a good few examples in general‑purpose media sound‑effects libraries, if you have anything like that to hand.

Published January 2010

Thursday, July 27, 2017

Q. How can I achieve a ‘dry’ sound?



By Various

I record and mix in my 'studio', which isn't too great acoustically. I can manage somehow when mixing, by working on headphones and doing lots of cross‑referencing, but the problem is that when it comes to recording I really hate the room sound on my vocals, and most of all on acoustic guitars, which I use a lot. The reverb tail is pretty short, but I'm still having a hard time getting a nice dry sound on my guitars, because I can't record dry! I know that the obvious solution is to treat the room, but the truth of the matter is that I can't do much better than this for now. So is there any way to treat a 'roomy' sound (on vocals and guitar) to make it sound drier? I know it is very difficult, or maybe impossible, especially for acoustic guitars, but any kind of suggestion, even for small improvements, would be very welcome.
 
A high‑resolution spectrum analyser such as Schwa's Schope lets you quickly and precisely home in on specific resonant frequencies that may be responsible for a coloured or uneven sound.

Via SOS web site

SOS contributor Mike Senior replies: Given that the reverb doesn't have a 'tail' as such, I reckon it's the reverb tone that's the biggest problem, so trying to use some kind of gating or expansion to remove it is unlikely to yield a useful improvement. You could help minimise the ambient sound pickup by using a directional mic for both vocals and guitar and keeping a fairly close placement. For vocals, very close miking is pretty commonplace, but for acoustic guitar you might want to experiment with using an XY pair of mics instead of a single cardioid, to avoid 'spotlighting' one small area of the guitar too much. That setup will usually give you a more balanced sound because its horizontal pickup is wider than a single cardioid on its own. In all but the smallest rooms, it's usually possible to get a respectable dry vocal sound just by hanging a couple of duvets behind the singer, and because I suspect that you've already tried this fairly common trick, I'm suspicious that room resonances are actually the biggest problem, rather than simple early reflections per se. Duvets are quite effective for mid‑range and high frequencies, but aren't too good at dealing with the lower‑frequency reflections that give rise to room resonances.

So given that room resonance is likely to be the problem, what can you do about it? Well, if you've no budget for acoustic treatment, I'd seriously consider doing your overdubs in a different room, if there's one available. If you're recording on a laptop, or have a portable recorder, maybe you can use that to record on location somewhere if you're confined to just the one room at home. I used to do this kind of thing a lot when I first started doing home recordings, carting around a mic, some headphones and a portable multitrack machine to wherever was available.

Part of what the room resonances will be doing is putting scary peaks and troughs into the lower mid‑range of your recorded frequency response, but the exact frequency balance you get will depend on exactly where your player and microphone are located in relation to the dimensions of the room, so a bit of determined experimentation in this respect might yield a more suitable sound, if not quite an uncoloured one. You might find that actually encouraging a few more high‑frequency early reflections using a couple of judiciously placed plywood boards might also improve the recorded room sound a little. A lot of domestic environments can have a bit too much high‑frequency absorption, on account of carpets, curtains, and soft furnishings.

After recording, you could also get busy with some narrow EQ peaks in the 100‑500Hz range, to try to flatten any obvious frequency anomalies. One thing to listen for in particular is any notes that seem to boom out more than others: a very narrow notch EQ aimed precisely at that note's fundamental frequency will probably help even things out. You can find these frequencies by ear in time‑honoured fashion by sweeping an EQ boost around, but in my experience a good spectrum analyser like Schwa's Schope plug‑in will let you achieve a better result in a fraction of the time. However, while EQ may address some of the frequency‑domain issues of the room sound, it won't stop resonant frequencies from sustaining longer, which is just as much part of the problem, and there's no processing I know of that will deal with that.

For my money, this is the kind of situation where you can spend ages fannying around with complicated processing to achieve only a moderate improvement, whereas nine times out of 10 you'll get better results much more quickly by just re‑recording the part.


Published January 2010

Wednesday, July 26, 2017

Q. What are filters and what do they do?


By Various
I always hear people talking about low‑pass filters and high‑pass filters and cutting at this and that frequency, but where do you get these filters from? I don't think I have one in my Cakewalk Project 5 software. Are they part of equalizers?
Via SOS web site

This diagram illustrates both low‑pass (high cut) and high‑pass (low‑cut) filtering. The shaded areas in the diagram will be attenuated. 
This diagram illustrates both low‑pass (high cut) and high‑pass (low‑cut) filtering. The shaded areas in the diagram will be attenuated.

SOS Technical Editor Hugh Robjohns replies: People sometimes use the terms 'EQ' and 'filter' interchangeably, so it's understandable that you might be confused. We've published several introductory guides to EQ, most recently in SOS December 2008 (/sos/dec08/articles/eq.htm), so if you're a bit baffled about the broader subject of EQ, it would be well worth reading this.

Essentially, EQ is used to boost or attenuate (turn down) a range of frequencies in order to shape a sound. High‑pass and low‑pass filters are common in professional equalisers, but less common in budget designs. They are used to define the highest and lowest frequencies of interest in the signal and they pretty much do what their names suggest: let audio above a certain frequency pass (high‑pass filter) or audio below a certain frequency pass (low‑pass filter). Anything outside those limits is attenuated. They are also called low‑cut or high‑cut filters, but the function is the same.

Filters are defined by their slope, which determines the attenuation of signals outside the 'pass' band. Most audio filters on mixing desks (and DAWs) will have a slope of 12dB or 18dB per octave, and in synthesizer filters the slope may be as steep as 24dB per octave. If an 18dB/octave high‑pass filter is set to 80Hz, any audio an octave below that (at 40Hz) will be attenuated by 18dB, and an octave lower still, at 20Hz, it will be attenuated by 36dB... and so on.

High‑ and low‑pass filters generally have much steeper slopes than the more normal equaliser bands (which are typically only 6dB/octave) and are intended for a different purpose. You can't effectively remove rumble with a bass EQ control, but you can with a high‑pass filter. But equally, you can't shape the tone of a bass guitar with a high‑pass filter as easily as you can with a bass EQ control.

Filters are used for 'corrective' equalisation, as opposed to creative equalisation. They are used to clean up a signal, rather than to shape the sound creatively. They only provide attenuation of unwanted frequencies, and there's no scope to boost any part of the frequency range. Of the two, the high‑pass filter is probably the most useful, as it helps to remove unwanted rumbles and other unwanted sub‑sonic rubbish that microphones tend to capture. Most DAW software includes a software EQ that you'll be able to use to perform any of these tasks, and although I'm not personally familiar with Cakewalk Project 5, I notice that it can host third‑party VST plug‑ins, so there are many freeware plug‑ins that you could use if your DAW doesn't have them built in.


Published July 2009