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.

Saturday, January 31, 2015

Q. Should I EQ my drum recordings?

When recording a drum part, do I have to EQ it, or should it be left untreated?




There's no hard-and-fast rule when it comes to treating your drum recordings (or not!). If you get a really good-quality source recording, you may not need to EQ it at all.There's no hard-and-fast rule when it comes to treating your drum recordings (or not!). If you get a really good-quality source recording, you may not need to EQ it at all.



Samuel Am, via email



SOS Reviews Editor Matt Houghton replies: There's no right or wrong answer to this question! As long as the drum sounds right to you when played through your studio monitors, use whatever you need to get the result you want: compression, EQ, Transient Designer or, in the case of a snare drum, perhaps even a tiny bit of distortion.



Back in the days of big-budget studio recordings, many engineers would — if they felt it was needed — EQ and compress drum mics on the mixing console while recording, and then print the results to multi-track tape for mixing. Today, it's easy to capture a clean recording and add EQ or other processing to taste at the mixing stage, using plug-ins. This leaves more options open to you, and avoids you getting stuck with poor EQ and dynamics decisions that don't suit the mix, but it also means that you're putting off decisions, so the whole process can take longer.



If you're new to this, I'd recommend experimenting first with mic choice and placement to get the best sound possible. Then add EQ and/or compression during the mix stage if you feel it's needed for the particular track in question. In the long run, you'll end up with better results if you start working in this way, and eventually your EQ and dynamics decisions will become second nature, at which point it will be easy to make them during recording rather than mixing.

There's no hard-and-fast rule when it comes to treating your drum recordings (or not!). If you get a really good-quality source recording, you may not need to EQ it at all.

Of course, depending on the style of the song, it's also perfectly possible to record a natural-sounding drum part with no processing whatsoever, just relying on mic choice and placement (and a good-sounding drum and room, and a good drummer!) to capture a clean sound.



As for which mics to pick, it really depends on the drum(s) you're using: what you use on a kick or snare close-mic can be very different from what you use for overheads or hand-drums.

   

Pa600 Professional Arranger - Performance by Marco Parisi

Friday, January 30, 2015

Q. How important are microphone self-noise and SPL figures?

Sound Advice : Miking



I am interested in the Shure SM7b mic and have been looking at its specifications, but the Shure web site seems to be missing information for self-noise and maximum SPL levels. I've heard people saying that the SM7 can handle up to 180dB SPL! I'm curious as to whether or not that is true (probably not) and if it is anywhere near that, I'm assuming it's because it's got some kind of -30dB switch on it or something crazy like that. Can you shed any light on this?



You won't find self-noise specifications for the Shure SM7b, as it is a dynamic (moving-coil) microphone. The only self-noise generated is the thermal noise from its own output impedance.You won't find self-noise specifications for the Shure SM7b, as it is a dynamic (moving-coil) microphone. The only self-noise generated is the thermal noise from its own output impedance.



Via SOS web site



SOS Technical Editor Hugh Robjohns replies: The reason you can't find those specific specifications is because the SM7 is a dynamic (moving-coil) microphone. In fact, you probably won't find those specs for any dynamic mic from any manufacturer (other than dynamic mics with built-in buffers or gain stages), because they are largely meaningless and pointless figures.



The self-noise generated by a moving-coil microphone is only the thermal noise generated by the mic's own output impedance, which is essentially just the DC resistance of the moving coil itself, plus that of a humbucking coil (if employed) and the output transformer (if present). This noise contribution is negligible, and will be utterly swamped by the receiving preamp's own electronic noise.

You won't find self-noise specifications for the Shure SM7b, as it is a dynamic (moving-coil) microphone. The only self-noise generated is the thermal noise from its own output impedance.

The maximum SPL level for a dynamic mic is determined mainly by the range of mechanical movement afforded to the coil, and that will be more than high enough for any conventional application. So it's not unusual to find professional dynamic mics that are capable of over 150dB SPL (for one percent THD), albeit with rapidly increasing distortion towards the limits, and with mechanical clipping occurring when the diaphragm and/or coil hits the end stops at 170 or 180 dB SPL.



In contrast, the self-noise and maximum SPL figures are quoted for all electrostatic mics (capacitor and electret) because the impedance converter electronics built into the microphone determine the mic's dynamic range capability, the lower limit being set by the amplifier's self-noise, and the upper limit by the amplifier's distortion or clipping.    

Korg All Access: Adam Deitch and WAVEDRUM Mini

Q. Can you recommend a low-cost heavy-duty mic stand?

Sound Advice : Miking




I have the usual selection of Stagg and anonymous mic stands, which are fine most of the time, but I now have some mics that are really pretty heavy (SE Electronics' Gemini III, for instance) and none of my present stands really cut it. Of course, all mic stands are described as 'heavy duty', but I'm looking for something that can hold really heavy microphones reliably and with the minimum of hard twisting of small knobs and so on.Of course, SE make a suitable stand, but I'm not sure I could justify $500 on one mic stand. Can you suggest anything usable below, say, $150?



An expensive mic stand might seem like a waste of money, given that most still suffer from 'droop', but some very well-engineered stands exist that do not suffer from this problem. This stand from Sontronics, for example, is more than worth its cost, given that it is protecting far more valuable mics that could last you a lifetime if well looked after.An expensive mic stand might seem like a waste of money, given that most still suffer from 'droop', but some very well-engineered stands exist that do not suffer from this problem. This stand from Sontronics, for example, is more than worth its cost, given that it is protecting far more valuable mics that could last you a lifetime if well looked after.



Via SOS web site

An expensive mic stand might seem like a waste of money, given that most still suffer from 'droop', but some very well-engineered stands exist that do not suffer from this problem. This stand from Sontronics, for example, is more than worth its cost, given that it is protecting far more valuable mics that could last you a lifetime if well looked after.

SOS Technical Editor Hugh Robjohns replies: If you can use a mic stand without a boom arm — so, just the vertical pole — there shouldn't be any problem, because even budget mic stands should be able to support the heaviest microphone without too much trouble. The real problem comes when trying to hang a heavy mic on a boom arm, because most ordinary mic stands don't have anything like a sufficient counterweight mass to properly balance even moderate mics, let alone big, heavy ones. As a result, the boom arm clutch has to resist almost all of the rotational force created by the leverage of the heavy mic at the end of the boom and, frankly, most just aren't up to the job. The inevitable consequence is the annoying 'droopage', and the more you try to tighten the clutch to prevent it, the quicker the whole thing wears out (or breaks), and quickly becomes droopy even when supporting light microphones!



The correct engineering solution is to properly counterbalance the weight of the microphone so that there is no net rotational force at the boom clutch. That then allows the clutch to do what it was intended to do — stop the boom arm from moving — rather than have to accommodate the entire rotational leverage. The cheap and cheerful solution is to tape or affix some additional weight to the end of the boom arm; you need enough to balance your heaviest mic at the maximum boom extension you plan to use. However, this will be ugly and may not be as safe as it should be, and you certainly don't want the weight to fall off onto someone's foot... or the mic to crash onto the floor shortly afterwards!



I know the idea of spending $500 on a mic stand seems silly, but, to be honest, I think it's worth it for peace of mind when you're working with mics that cost $1500 and potential personal injury insurance claims! Moreover, mic stands in this cost bracket generally live forever, because they are so well designed and rugged, which means that the amortised investment is actually very low.



The SE mic stand is surprisingly stable, but it is a kind of hybrid of a reverse-engineered Keith Monks boom arm and clutch from the 1970s and a drummer's cymbal stand. It does have a heavier counter-weight than most budget stands, but it's still not an ideal solution, to my mind.



The most cost-effective and properly engineered stand I've come across to date is the Sontronix Matrix 10. It's not the prettiest or most compact stand on the planet — it's basically a modified photography lighting stand — but it has cogged clutches that definitely won't slip, a very sensible counterweight, removable wheels, and a handy drop-arm. It's very secure, totally reliable, and there's nothing to break, so it will live forever. I reviewed it in the August 2010 edition of Sound On Sound (see the full review at /sos/aug10/articles/sontronics-matrix-10.htm).



If you want something in matt black and with a much smaller footprint, I've just been reviewing the Latch Lake MicKing stands, which I have to say are utterly brilliant. However, they are also pretty expensive, because they are very well engineered, and imported from the US. The review is soon to appear in Sound On Sound, but these stands have a sensibly massive counterweight on the boom arm, a very heavy, but compact, base (with transport wheels to make it easy to move the stand to a storage area), a nice drop-arm system, and really ingenious lever locks and clutches that are adjustable for both tension and ease of use. These are very solid and impressive stands and well worth the investment, in my view.    

Thursday, January 29, 2015

Korg TM-50 (Tuner/Metronome) and Korg TMR-50 (Tuner/Metronome/Recorder) Product Overview

Q. Can I use my aux send/return loop, or do I need insert points?

Sound Advice : Mixing




I'm trying to hook a Behringer Denoiser and an SPL Vitalizer MK1 into an older-model Alesis Multimix 16USB mixing console, working with the sends and returns. I purchased two sets of send/Y-leads, which are obviously TRS single-to-dual monos, since the desk is a single jack send, but double on the returns. Now, when I'm fully set up on both units, and I plug in fully, I am missing one channel on each. I'm using the less reliable method of inserting the Y-lead plug halfway into the insert until there is a springy 'click' feeling and all is fine. Do I need to purchase a different-style lead, a TS, and not a TRS? I've not used many outboard effects before in send modes, but am digging out old gear that may still be of use.



If your console doesn't have channel-insert sockets and you want to be able to process individual source channels, the easiest solution would be to use a patchbay. If your console doesn't have channel-insert sockets and you want to be able to process individual source channels, the easiest solution would be to use a patchbay.



Via SOS web site

If your console doesn't have channel-insert sockets and you want to be able to process individual source channels, the easiest solution would be to use a patchbay.

SOS Technical Editor Hugh Robjohns replies: Hooking up the Behringer Denoiser and the SPL Vitalizer MK1 into the Alesis Multimix should be easy enough. The first port of call is the manual, to check how the mixer is wired. In this case, there are no channel or mix insert points, but there are two mono aux sends (called Aux A and Aux B), and both are wired as impedance-balanced outputs. That means that there is a signal on the tip connection, but no signal on the ring connection.



The mixer also has two stereo balanced effects returns (called FX Return A and B), although the B return socket is normalled from the output of the internal FX processor. Stereo FX Return A is wired such that if you only plug something into the left channel, it is normalled across to the right as well. FX Return B does not have that facility. As I recall, the original stereo Vitalizer had both balanced XLRs and unbalanced quarter-inch TS sockets for the inputs and outputs, and the Behringer SNR2000 two-channel Denoiser has both XLR and TRS sockets on the inputs and outputs, wired balanced, but usable unbalanced.



Given these interconnection formats, it makes sense that you'd be missing one channel with your types of cables. The kind of cable you have connects the TRS tip to the tip of one of the TS plugs, and the TRS ring to the tip of the other. With an impedance-balanced output, there will, therefore, only be a signal on one of the TS plugs.



That's physically what is available, so how should things be connected? Firstly, in terms of the send/return Y-leads you've purchased, I think you've misunderstood what's going on here. Each aux output is a mono send. The effects returns are stereo returns. This is quite normal because, typically, you'd be patching a stereo reverb across them: taking a mono input to the reverb and creating a stereo return signal, for example.



Both the Vitalizer and the Denoiser are stereo or dual-channel devices — so what are you trying to achieve? If you want to process the main mix bus, the aux send/effects return loop can't access the mix bus at all. If you want to process a stereo input channel, the aux sends are both derived mono sums, so that won't work in stereo either. And, if you want to process the input channels individually in mono, plugging up both sides of both processors is pointless.



But, fundamentally, both the Vitalizer and the Denoiser are really insert processors, not send-return processors. They are both designed to work directly on the source signal, and the processed signal is then mixed with all the other console inputs. Neither the Vitalizer nor the Denoiser generates an independent return signal — like, say, a reverb does — that you would want to mix alongside everything else. Basically, these tools are simply not designed to be used in an aux-send/effects-return configuration.



If you want to be able to process individual source channels through the Vitalizer or Denoiser, and the console doesn't have channel-insert sockets (and yours doesn't), the easiest solution would be to invest in a TRS patchbay. You could then manually patch the source signals either directly to the mixer inputs, or to a processor input, and then patch the processor output back to the appropriate mixer input. You could even patch the stereo mix out via the Vitalizer or Denoiser before sending it on to your recording and monitoring chain. That would be a far more practical and sensible solution.    

MicroMadness Video Contest Worldwide Winners!

Wednesday, January 28, 2015

Q. Which budget tube preamp should I buy?

I'm looking to get a tube preamp but am on quite a small budget. I've already looked at the ART Tube MP and PreSonus TubePRE. Can anyone offer any opinions on these, or suggest any other potential buys? My budget is around $150, although I may be going halves with a friend, so would welcome suggestions up to around $250.



If you only have a small budget, a tube preamp may not be the best way to go. If you can stretch to more than few hundred pounds, the TL Audio Ivory, for example, could be a good buy. Otherwise, the preamps you may already have on a decent audio interface are hard to beat.If you only have a small budget, a tube preamp may not be the best way to go. If you can stretch to more than few hundred pounds, the TL Audio Ivory, for example, could be a good buy. Otherwise, the preamps you may already have on a decent audio interface are hard to beat.



Via SOS web site

If you only have a small budget, a tube preamp may not be the best way to go. If you can stretch to more than few hundred pounds, the TL Audio Ivory, for example, could be a good buy. Otherwise, the preamps you may already have on a decent audio interface are hard to beat.

SOS Review Editor Matt Houghton replies: This question crops up quite a lot. The first thing to ascertain is why you want a tube preamp, because, in my experience, most people who ask this question haven't really thought about it, and generally haven't arrived at any answer. So what qualities are you seeking from a tube preamp? Warmth? Flair? Clarity?



Personally, I've used a number of budget and high-end tube preamps, and there are very few 'affordable' ones that I'd countenance, simply because there are far, far better solid-state preamps at the same price. I'd count anything below $700 in that category. Between $700 and $1500 there are a few that may be worth a look if they do what you want: the SPL Goldmike or the TL Audio Ivory/Ebony, for example.



But if you're looking for 'warmth', you're arguably more likely to get that from a preamp or other device with audio transformers. If it's 'flair', you might find that picking the right mic (which may very well be a tube mic) is a better option.



At your sort of budget, the bang-for-buck you get from the preamps on a decent audio interface is hard to beat, in my opinion. Going up to about $400 starts to bring the likes of the GA Pre73 into view. But then $400 might very well buy you a nicer mic that does what you want!    

Korg Krome Video Manual -- Part 5: Global Mode & Media Mode

Q. Should I apply an effect to the whole mix, or use effects on each track?

Sound Advice : Mixing



Does sending multiple instruments to the same effects track (reverb/delay) have a different effect on the overall mix from applying effects to each individual channel? What is the best way to apply lots of effects to tracks whilst saving CPU?Also, how does this work in terms of mixing multiple tracks through one compression effects track?



Via SOS web site



SOS Reviews Editor Matt Houghton replies: If I had a pound for every time I've been asked this question I'd be a rich man!

An efficient way to route effects is to send multiple channels to the same effects processor.

Let's start with reverb and delay: I'll discuss reverb, but the same principle applies to delays. It's worth pointing out that only recently, with plug-in-based DAWs and powerful computers, has it really become possible — at least without being a millionaire — to run separate reverbs on every channel. In the days of yore, almost all reverbs would have been set up as send effects. Partly, that was down to the cost, but there were some practical and sonic benefits to this approach too. Firstly, using reverb as a send effect meant it was quick and easy to adjust the wet/dry balance of things, either using the aux send knob on each source channel, or using the fader on the effects return channel, depending on whether you wanted to adjust the balance of one source or the whole mix. Secondly, you could leave reverbs patched into aux sends, so that your favourite reverb unit was always easily accessible to each channel, without needing to re-patch. Thirdly, if the aim is to gel different sources together, you probably want them to sound like they're in the same space, so it makes sense to share the same reverb. And, finally, separating the reverb and the source track allows you to further process the reverb signal, with filters or other processors, before mixing it back with the source.An efficient way to route effects is to send multiple channels to the same effects processor.An efficient way to route effects is to send multiple channels to the same effects processor.



However, it's also quite common for some sources to have their own dedicated reverb. A snare drum, for example, might be sent both to an overall ambience or drum-room reverb, as well as to its own dedicated reverb. In this situation, you could use a reverb as an insert or as a send effect, and if you wanted the snare reverb to be processed along with the snare (or the rest of a drum group), you'd route the snare reverb's return channel to a group/bus track along with the snare, or all of the drums. Another possibility is that you simply want to thicken up a weedy sound using a short ambience patch and, again, you can use either approach here. So, in short, it's perfectly acceptable to use a reverb either as an insert or as a send, but the send approach is much more common, and for good reason.



When it comes to EQ and dynamics processors, such as compressors, gates and limiters, this situation is reversed. In other words, they'd normally be used as inserts, to sculpt sounds and manage their dynamic range so that they better fit the mix (which is why some high-end consoles feature EQ and compression on every channel). Again, though, there are occasions when you might want to use them as sends, the most common of which is when you want to perform parallel compression. Some compressors feature a wet/dry, or 'blend', control for exactly this purpose, but you have more flexibility by using a compressor as a send effect, because you're able to EQ the return signal again before blending back with the source. This technique is fairly commonly deployed on a drum bus, sometimes with a slight 'smile' EQ in series with the compressor. If you want another, more extreme example, producer Michael Brauer reportedly mults out vocals to several different compressors in parallel to get the effect he wants.



Of course, it's also perfectly acceptable to use processors such as compressors on bus channels without doing any parallel compression. That's known as 'bus compression' and is typically done to 'glue' or 'gel' things together. You'll often see that done on a drum group bus or on a master bus, and if that's a technique that you're interested in I'd recommend reading the feature in SOS May 2008 (/sos/may08/articles/mixcompression.htm).



If you're working in a digital environment, like a modern computer-based DAW, it's worth a quick word about latency compensation. Most DAWs now include automatic plug-in delay compensation, and this is essential when doing parallel compression, as the delays will otherwise cause unwanted phase-cancellation. It's not really an issue for reverbs, though, where any delay can be compensated for by reducing the reverb's pre-delay.    


Tuesday, January 27, 2015

Korg Krome Video Manual -- Part 4: Sequencer Mode

Q. How can I warm up my recording without using EQ?

Sound Advice : Recording




I've put a lot of effort into creating and editing a recording of solo mandolin — played quite slowly — and although I like the final result a lot, on consideration the tone is too trebly and cold, almost like a photograph with too sharp a resolution. A friend mentioned he thought I could perhaps 'warm it up' using compression, perhaps of a type designed for vocals. Can you give me some guidance on how best I might do this? Of course, I realise I can use EQ, but would specifically be interested in any thoughts on how compression/limiting could be used on an existing take to get a warmer result. I've used Logic and the recording is clear, undistorted, and free from ambient sound.



Simon Evans via email

The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.

SOS contributor Mike Senior replies: There are ways to warm up a mandolin sound subjectively using compression, although none of them are likely to make as big an impact as EQ. Fast compression may be able to take some of the edge off a mandolin's apparent tone, for instance, assuming the processing can duck the picking transients independently of the note-sustain elements. There are two main challenges in setting that up. Firstly you need to have a compressor that will react sufficiently quickly to the front edges of the pick transients, so something with a fast attack time makes sense. Not all of Logic's built-in compressor models are well-suited to this application, so be sure to compare them when configuring this effect; instinctively I'd head for the Class A or FET models, but it's always going to be a bit 'suck it and see'. The second difficulty will be getting the compressor not to interfere with the rest of the sound. The release-time setting will be crucial here: it needs to be fast enough to avoid pumping artifacts, but not so fast that it starts distorting anything in conjunction with the attack setting. Automating this compressor's threshold level may be necessary if there are lots of dynamic changes in the track, for similar reasons. Applying some high-pass filtering to the compressor's side-chain (open the Logic Compressor plug-in's advanced settings to access side-chain EQ, and select the 'HP' mode) may help too, because the picking transients will be richer in HF energy than the mandolin's basic tone.The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.



Another way to apparently warm up a mandolin is to take the opposite approach: emphasise its sustain character directly while leaving the pick spikes alone. In a normal insert-processing scheme, I'd use a fast-release, low-threshold, low-ratio (1.2:1 to 1.5:1) setting to squish the overall dynamic range. Beyond deciding on the amount of gain reduction, my biggest concern here would be choosing an attack time that avoided any unwanted loss of picking definition. In this case, shelving a bit of the high end out of the compression side-chain might make a certain amount of sense if you can't get the extra sustain you want without an unacceptable impact on the picking transients.



Alternatively, you might consider switching over to a parallel processing setup, whereby you feed a compressor as a send effect, and then set it to more aggressively smooth out all the transients. The resulting 'sustain-only' signal can then be added to the unprocessed signal to taste, as long as you've got your plug-in delay compensation active to prevent processing delays from causing destructive phase-cancellation. Using an analogue-modelled compressor in this role might also play further into your hands here, as analogue compressors do sometimes dull the high end of the signal significantly if they're driven reasonably hard, giving you, in effect, a kind of free EQ.  


Korg Krome Video Manual -- Part 3: Combination Mode & Effects

Monday, January 26, 2015

Q. Are some analogue signal graphs misleading?

Sound Advice : Mixing




I read your feature about 'Digital Problems, Practical Solutions' (/sos/feb08/articles/digitalaudio.htm), which said that digital audio can capture and recreate analogue signals accurately, and that the 'steps' on most teaching diagrams are misleading. Does that mean that the graph should really show lines, or plot 'x's, instead of looking like a standard bar-graph?



Remi Johnson via email



SOS Technical Editor Hugh Robjohns replies: Good question! The graphs in that article are accurate as far as they go, but offer a very simplified view of only one part of the whole, much more complex, process.



When an analogue signal (the red line on Graph 1: Sample & Hold) is sampled, an electronic circuit detects the signal voltage at a specific moment in time (the sampling instant) and then holds that voltage as constant as it can until the next sampling instant. During that holding period the quantising circuitry works out which binary number represents the measured sample voltage. This, not surprisingly, is called a 'sample and hold' process, and that's what that diagram is trying to illustrate. Graph 1: Sample & HoldGraph 1: Sample & Hold

Graph 1: Sample & Hold

So the sampling moment is, theoretically, an instant in time, best represented on the graph as a thin vertical line at the sample intervals (the blue lines in the picture Graph 1: Sample & Hold), but the actual output of the sample and hold process is the grey bar extending to the right of the blue line.



However, the key to understanding sampling is understanding the maths behind that theoretical sampling 'instant', and that means delving into the maths of 'sinc' (sin(x)/x) functions, which is the time-domain response of a band-limited signal sample. At this point most musicians' eyes glaze over…

Graph 2: Two Sinc Functions

As we know, the measured amplitude of each sample from an analogue waveform is represented by a binary number in the digital audio system. When reconstructing the analogue waveform that number determines the height of the sinc function.



The important point is that we are not just creating a simple 'pulse' of audio at the sample point, because the sinc signal actually comprises a main sinusoidal peak at the sampling instant (and of the required amplitude), plus decaying sine wave 'ripples' that extend (theoretically for ever) both before and after that central pulse. The reconstructed analogue waveform is the sum of all the sinc functions for all the samples.

Graph 3: 3kHz Sinc Addition

The clever bit is that the points where those decaying sinc ripples cross the zero line always occur at the adjacent sampling instants. This is shown in the next diagram (Graph 2: Two Sinc Functions) where, for simplicity, just two sample sinc functions are shown for samples 23 (red) and 27 (blue). You can see that at the intermediate sample points (26, 25, 24 and so on) the sinc functions are always zero.Graph 2: Two Sinc FunctionsGraph 2: Two Sinc Functions



That means that the ripples don't contribute to the amplitude of any other sample, but they do contribute to the amplitude of the reconstructed signal in between the samples, with the adjacent sample sinc functions having the greatest influence, and lesser contributions from the more distant samples. This is shown in the next diagram (Graph 3: 3kHz Sinc Addition), in which the sinc functions of a number of adjacent samples are shown, and when summed together produce the dotted line that is a sampled 3kHz sine waveform. Graph 3: 3kHz Sinc AdditionGraph 3: 3kHz Sinc Addition



These last two diagrams have been borrowed from a superb paper by Dan Lavry (of Lavry Engineering), which explains sampling theory extremely well, and can be found here:
www.lavryengineering.com/documents/Sampling_Theory.pdf.    

Korg Krome Video Manual -- Part 2: Program Mode

Q. How do I record a hammered dulcimer?

Sound Advice : Recording



I've offered to have a go at recording a couple of friends of mine who play as a duo. For some of the songs, one of them will be playing a hammered dulcimer, which is something I've never tried to record before — do you have any advice?


Carl Turner


SOS Editor In Chief Paul White replies: Ideally, you need a stereo pair to give the sound some width and to help keep the volumes even across the strings. I usually place a pair of small-diaphragm, cardioid capacitor mics above the instrument, spaced by about two-thirds the instrument's width and around 500mm above the strings. Large-diaphragm models also work fine. You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.



You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.
You need to keep other sounds away, though, otherwise you'll pick up a lot of spill; you can't mic much closer without upsetting the string balance and changing the tone. Spaced omnis would also work if the recording is an overdub being done in a sympathetic room where spill is not an issue. I know some players use contact mics for live work, so if you are recording at the same time as other loud instruments, it could help to take a feed from a pickup as well, just in case the overhead mics pick up too much spill. You may even be able to blend the contact mic and the overheads, as long as you compensate for any phase problems that might arise by time-aligning the waveforms and checking their polarity in your DAW.    


Saturday, January 24, 2015

Korg Krome Video Manual -- Part 1: Introduction and Navigatio

Q. How can I even out the bass response in my room?

Sound Advice : Recording




The bass response in my room is quite variable, with some notes appearing boomy and others being quite weak, but the overall amount of bass seems about right. I've been told that I need to install bass traps to cure the 'lumpiness', but won't that just remove bass from the room rather than smooth it out?



Via SOS web site

Bass traps don't 'remove' bass from a room; instead, they should support the bass that is produced by the speakers, often making it louder and more consistent.

SOS Technical Editor Hugh Robjohns replies: It's a very common fallacy to think that bass traps will remove bass, and the way bass traps work isn't entirely intuitive. But, in fact, effective bass trapping allows the room to support all of the bass that the speakers produce, so you'll normally get more bass, overall, not less. Bass traps don't 'remove' bass from a room; instead, they should support the bass that is produced by the speakers, often making it louder and more consistent.Bass traps don't 'remove' bass from a room; instead, they should support the bass that is produced by the speakers, often making it louder and more consistent.



In a room without effective bass trapping, the low frequencies produced by your speakers head out into the room and reflect back off the walls and other boundary surfaces. Since low frequencies have very long wavelengths (in the order of many metres) and most home studios are in relatively small rooms, the reflected low‑frequency sound meets the direct bass still coming from the speaker. The phase relationship between the two is, therefore, critically important, because if they meet in nearly opposite polarities, they'll cancel each other out to some extent, and you'll end up with a significant dip in level. This is why some bass notes are often missing or very weak. On the other hand, if the direct and reflected sound waves meet nearly in the same polarity, they'll add and you'll get a peak in level and a boomy note. The different length, width and height dimensions of the room will determine the specific wavelengths and/or frequencies of the boosts and dips in level.



The idea of bass trapping is to soak up the low‑frequency sound-wave energy at the room boundary as it reaches and enters the trap. With that sound energy absorbed, it can't reflect back into the room, and so it can't interfere with the wanted direct sound from the speaker. The result is that you hear only the bass that the speaker produces and not the room effects, and that generally means that you end up with more bass and, more importantly, a substantially more even bass response both at the listening position and throughout the room.



However, the practical problems associated with effective bass trapping are that, firstly, simple absorptive bass trapping has to be physically deep so that it 'sees' a significant portion of the low‑frequency sound wavelength, and that can be difficult to integrate into a small room because of the space it takes up. The common alternative is to use some form of tuned resonant trap but, because that is inherently only effective over a very narrow band of frequencies, it requires sophisticated acoustic measuring tools to design and optimise correctly for the specific situation, and so it can be expensive.



Nevertheless, any amount of bass trapping is always helpful; unlike broadband absorbers, it's practically impossible to overdo it; and it certainly won't remove bass from your room!    

Friday, January 23, 2015

Korg Pa600 Video Manual -- Part 7: Global, Media, and Updates

Q. Why has my mixer socket gone intermittent?

My Mackie VLZ mixer's got a bad socket: when the mixer is switched on, the socket goes on and off intermittently. If it's just a loose connection inside, I think maybe I could fix it, although I'm only an enthusiast when it comes to electronic wiring. Do you have any ideas?




Via SOS web site
Cleaning the sockets on your mixer (or any other piece of equipment, for that matter) should be relatively simple. Using a slightly 'roughed‑up' jack with some Deoxit and plugging and unplugging the jack repeatedly should get rid of any oxidation or debris that's interfering with connections.


SOS Reviews Editor Matt Houghton replies: I've owned quite a few VLZ Pro series mixers over the years, as I've found that they're ridiculously good value for a rack of preamps of that quality, particularly when I've bought second‑hand. Of course, buying second‑hand means that I've encountered quite a few crackly and intermittent channels, too! In my experience, usually, these symptoms are down to the insert jack sockets becoming a bit dusty or grimy, which results in a bad contact when a jack is put in there, and can fool the mixer into thinking you've inserted something when you haven't! Cleaning the sockets on your mixer (or any other piece of equipment, for that matter) should be relatively simple. Using a slightly 'roughed‑up' jack with some Deoxit and plugging and unplugging the jack repeatedly should get rid of any oxidation or debris that's interfering with connections.Cleaning the sockets on your mixer (or any other piece of equipment, for that matter) should be relatively simple. Using a slightly 'roughed‑up' jack with some Deoxit and plugging and unplugging the jack repeatedly should get rid of any oxidation or debris that's interfering with connections.Q. Why has my mixer socket gone intermittent?



If this is the case with your mixer, all you need do is give the insert socket a good clean. In fact, you might as well clean them all anyway, to prevent this issue occurring on other channels. A spray of Deoxit (which is slightly different from, and better than most, 'contact cleaners') and a very lightly 'roughed up' jack plug are all you need. By roughed up, I mean that it just needs to be lightly scratched with a wire brush or coarse sandpaper (don't overdo it!), so that it can lightly scratch the socket when placed inside. Squirt some Deoxit into the offending socket, then plug the jack plug in and out of the socket 10 or 20 times, or so. That should get rid of any muck and oxidised surfaces, ensuring a good clean contact when in use, and that there are no problems when it's left empty.

Q. Why has my mixer socket gone intermittent?

If that isn't the issue, it's hard to say precisely what is without examining the mixer in question. It could be the state of the input sockets, or of any of the pots and switches. It could be trickier to solve such problems, since the Mackie not a modular desk, and that means that it will take time to open it up, trace the problem, fix it and put everything back together again. In fact, I'd venture to suggest that it might be cheaper to buy a replacement second‑hand version of the same mixer than it would be to get it serviced!  


Korg Pa600 Video Manual -- Part 6: Recording a Song

Thursday, January 22, 2015

Q. How should I treat my sample-library orchestral parts?

Sound Advice : Mixing



I use a lot of orchestral parts (EastWest Quantum Leap Symphonic Orchestra) in my contemporary songs, and I was wondering whether I should I mix them in precisely the same way as the other tracks in my arrangement (in other words, mix and EQ each instrument of the orchestra separately on its own separate track), or whether I should treat them as an ensemble to try and maintain some sort of sonic integrity? Also, is there any reason to be precious about using stereo samples and/or the natural reverb that is supplied via the stage/surround mics? Or will dry tracks plus a good‑quality plug‑in reverb do just as well? And how closely should I stick to the standard orchestra layout in terms of panning? I realise these are partly subjective decisions, but I just wondered whether there might be any technical arguments that would help me make a start.



Garfield Mayor via email



SOS contributor Mike Senior replies: The main technical consideration to bear in mind is that samples designed for creating orchestral scores (such as the EastWest Quantum Leap library) aren't really intended to be layered in the background of a chart production. As such, they're likely to have more low end than you need, bringing with it a risk of overall tonal muddiness and/or not enough high‑end sizzle to compete with pop‑style sounds, which are often brighter and more aggressive. That isn't a deal‑breaker, though, because it's mostly just a question of not being shy with the EQ if you need to reshape them into the context of your specific mix, much as you would any other track in your production. You might also want to use a little more compression than you'd typically expect in scoring work, because classical parts tend to have a dynamic range that's wider than can be accommodated in most chart work.



As far as panning is concerned, you could simply leave it as it's presented by default in the library itself, reflecting the natural placement of the instruments within the orchestra, but there are also a lot of good reasons why you might want to deviate from this standard setup on occasion. For example, if a bass instrument (whether melodic or percussive) is making an important low‑end contribution to the mix, there's a good argument for centring it for the purposes of playback on smaller stereo systems, as this will tend to maximise their low‑frequency playback efficiency. Also, if there's any important melody being carried by a single orchestral instrument/section, you may wish to centre that also, much as you might a guitar solo, to focus the listener's attention on it and help it to translate strongly for anyone listening in mono or on one side of a pair of shared earbuds. In my opinion, only about one listener in a hundred is likely to give a tinker's cuss about whether the panning is in any way 'real', they just want to be able to hear the music!



I would qualify that, however, by saying that you will almost certainly want a reasonable width out of the orchestral picture nonetheless, because the normal purpose of orchestral instruments in chart tracks is to make the sound more expansive. For this reason, I probably would use the stereo samples, if possible, because they'll give a richer and slightly more diffuse and enveloping sound in stereo.



If the EastWest Quantum Leap reverb/room tracks are placing undue strain on your computer, I wouldn't think twice about ditching them and replacing them with plug‑in reverb. It's not some kind of purist classical situation, so you can afford to cut some corners in the name of convenience. In fact, you may find that being able to EQ (and maybe even gate) the orchestral reverb plug‑in's return channel may help a good deal to avoid mix clutter. It does probably make sense to try to make the whole ensemble sound like it belongs together most of the time (and shared reverb effects will help there), simply because that's what most people are expecting. But if that conflicts with the needs of the balance, don't be afraid of treating some individual instruments to separate them from the group, especially where there are important instrumental solos. In fact, you'll hear this happening in a lot of film/TV scores if you listen out for it, as well as some very odd balances; a low‑register flute note audible against trombones, for instance, or harps right up front amidst an orchestra in full tantrum mode.    

Korg Pa600 Video Manual -- Part 5: Songbook

Q. Is it possible to customise icons in Logic?

I've always liked Logic's track-icon facility, as having a graphical representation of what sound or instrument is on any particular track makes navigating around the arrange page very fast and intuitive; in fact, I only wish Logic also displayed these same icons on the mixer page. Of course, Logic comes with a palette of useful icons, which are all well and good, but sometimes there isn't a suitable icon for what I'm working on. There are some notable omissions, such as some individual drum kit parts, which I'd like to be able to use when my drums are spread over several tracks. There's a steel drum, for example, but no separate snare drum or toms. How can I increase the usability of my icons?

Alhough you can already select a great number of icons to use on your tracks in Logic, you can also make your own, for more unusual instruments — as Paul White did for his eBow!


Chris Quayle via email



SOS Editor In Chief Paul White responds: You can change the icons easily enough from the default icon in the Inspector on the left‑hand side of the Arrange page, but because of the limitations you mention above, I like to create my own.



You can do this pretty easily by using photos or a graphics package, but the resulting icon size must be no more than 128 x 128 pixels, and it will look best if saved with a transparent background. However, if you're not artistically obsessive‑compulsive, a plain white background also looks OK and is much easier to achieve. Logic's icons are all saved in PNG format, with individual icons identified by number rather than name. The number determines in what order the icons will appear, so you can either replace an existing Logic icon or tack your new ones onto the end by giving them numbers that start after Logic's own icons finish. Currently, Logic has 325 icons, so your first new icon would be named 326.png.



A very simple way to create your own icon from a photograph is to use iPhoto, which is something every modern Mac has. In this example, I used a photograph of my eBow taken on a piece of white card, then added it to my iPhoto library. Using the Edit menu in iPhoto, I cropped the picture (making it look reasonably square) and then selected Export from the File menu. In the Export box I set the file type to PNG and the size to 'Custom', with a maximum size setting of 128 pixels. Once the exported picture was saved, I renamed it with the next available number and added it to Logic's own list of icons. You can use this simple method to add your band-member portraits to the list, as well as any unusual instruments or devices.Alhough you can already select a great number of icons to use on your tracks in Logic, you can also make your own, for more unusual instruments — as Paul White did for his eBow!Alhough you can already select a great number of icons to use on your tracks in Logic, you can also make your own, for more unusual instruments — as Paul White did for his eBow!



If life seems too short to be creating your own icons, there are several free sets available to download from various sites: just do a search for 'Logic Pro Icons' and several will pop up, offering pictures of guitars and countless synths, as well as a few more off‑the‑wall images. If you do fancy creating your own images, consider adding them to one of the online libraries so that other users can benefit from them.



The next trick is finding the right place to put your new icon files, as it isn't immediately obvious:



Go into the Applications folder and find Logic Pro.

Control-click Logic Pro to bring up a menu, and select 'Show Package Contents'.

Go to Contents / Resources / Images / Icons, and you'll see all the existing icons neatly numbered.



If you have the patience, you could renumber them to get them appearing in a new order (or replace some with your own), but I'm guessing that most users will simply add their own icons to the list starting at the next available number.    

Wednesday, January 21, 2015

Korg Pa600 Video Manual -- Part 4: Song Play

Q. What’s wrong with wall-warts?

I often read in your gear reviews that you don’t like ‘wall-warts’ and that good power supplies should feature a toroidal transformer. What’s wrong with ‘wall‑warts’ or in‑line lumps? Is it just convenience and the tangle of cables and plugs that concerns you, or is there a performance reason? And why is a toroid any better than any other power transformer? I’ve got different bits of gear that use different PSUs like this and they all seem to work!

Dylan Ashfield via email
SOS Technical Editor Hugh Robjohns replies: You’re absolutely right, I don’t like wall‑wart power supplies, but I don’t think I’ve ever advocated total exclusivity for toroidal transformers!
The problem with wall‑wart supplies, for me, is mainly a practical one. They easily become divorced from the product to which they belong, and it’s often difficult to match up the correct wall‑wart with the correct product if you have a multitude of wall‑wart-powered units. Then there’s the problem of plugging them in: most are larger than a normal 13A plug top, meaning that adjacent sockets become unusable. They are also often quite heavy and can work loose from vertical sockets. ‘In‑line lumps’ aren’t ideal, either, but they are usually a little more manageable than wall‑warts.
The worst thing about wall‑warts is that they are cumbersome. However, they are sometimes a necessity — for example, in the case of sensitive equipment that could suffer interference from transformers.

The worst thing about wall‑warts is that they are cumbersome. However, they are sometimes a necessity — for example, in the case of sensitive equipment that could suffer interference from transformers.
From a practical perspective, I much prefer products that have built‑in mains power supplies, so that the only power connection is a standard IEC cable. It’s far easier to manage, easy to carry spare cables, and much neater to install. However, I recognise that an external power unit is sometimes a necessity, such as in the case of sensitive mic preamps, for example (especially those with input and output transformers), where the inclusion of a mains transformer in the box can seriously degrade the performance through magnetically induced mains hums.
As for the type of power supply, linear supplies with large tranformers are relatively simple to engineer, and these often employ toroidal transformers because they are more compact and radiate a smaller external magnetic field than most laminated-core types. The down side is that linear power supplies are generally heavy, relatively large, and potentially quite inefficient. ‘Switched mode’ (SMPS) or ‘universal’ power supplies are far more complicated to design, but are small, lightweight, can cope with a wide range of input supply voltages, and can generate an array of precisely controlled output voltages. They also have negligible external magnetic fields, making them well suited for use in sensitive equipment like mic preamps!
As you say, a well‑designed power supply does what it says on the box, regardless of the specific technology used, but I think quality, reliability and user convenience should come above cheapness in the list of priorities!

Tuesday, January 20, 2015

Korg Pa600 Video Manual -- Part 3: Styles

Finding The Right PA System For You


Primer

Sound Advice : Miking




Thinking of taking your act to the stage? Then you'll need a suitable way of amplifying it, and that all depends on what and where you're playing...

Finding The Right PA System For You

Paul White



Finding The Right PA System For You



Most articles on PA systems start out by explaining how the various types of speaker cabinet work, but in this one I'm going to try to approach the subject from a different angle: looking at common types of act and venue, and then suggesting practical PA solutions for them.



These suggestions are based on the assumption that many of today's musicians travel by car rather than by van, and that they'd therefore like something as light and portable as possible that will still be able to do the job properly. Speaker systems have become more compact and more efficient over recent years, with the introduction of technologies such as switching-mode power supplies and Class-D amplifiers. This means that active speakers can be made to weigh very little more than their passive equivalents, while still being more convenient to set up, as there's less need for cabling. There's another advantage to active systems, in that suitable speaker protection can be, and usually is, built in.



A further advantage is that if each speaker has its own amp, you can muddle along on one speaker if the other fails. However, some small systems are based around two satellite speakers and a sub, where all the power amps are in the sub — so if the sub electronics fail, you lose your entire PA. This happened to me once while playing a small pub, but happily we were able to utilise the powered monitors as a substitute PA to get through the evening — it pays to always have a backup plan!

Going Solo?

For smaller gigs, acoustic instrument amplifiers will often serve well as compact PA systems, as many of them incorporate simple mixer functionality and can accommodate both mic and instrument inputs.

For smaller gigs, acoustic instrument amplifiers will often serve well as compact PA systems, as many of them incorporate simple mixer functionality and can accommodate both mic and instrument inputs.For smaller gigs, acoustic instrument amplifiers will often serve well as compact PA systems, as many of them incorporate simple mixer functionality and can accommodate both mic and instrument inputs.



Where the performer relies entirely on an instrument such as guitar or electric piano for accompaniment, rather than on backing tracks, a combo amp designed for acoustic guitar is often perfectly adequate for smaller venues. Unlike electric guitar amps, which are designed to impart their own sound, acoustic amps are essentially miniature PA systems, intended to simply amplify the signal going into them. They almost invariably offer an XLR mic input in addition to the instrument input, and so are ideal for singer/songwriter types. The speaker systems in such amps either use small full-range drivers or a driver teamed with a tweeter. They can all handle the vocal range perfectly adequately, and most also include some basic effects, such as reverb, making them good all-in-one solutions.



As long as they're not set up too high from the ground, combos can be used behind the performer at a reasonable level before feedback becomes a problem. The majority also include a pole-mount socket, facilitating mounting on a speaker stand.



I'd recommend a model of at least 70 Watts for small gigs, but do bear in mind that the amplifier power rating alone doesn't give an accurate indication of how loud the amplifier can go. The efficiency of the loudspeakers also plays a major part, so when you read the spec sheet, the SPL (Sound Pressure Level) is the most significant figure. To give you an idea of typical SPL figures, small combos are almost all capable of producing around 105dB (measured at one metre), but from a good full-size '12-inch plus horn' cabinet you could expect around 130dB, and sometimes even more.



AER are seen by many as the benchmark for compact acoustic guitar amplifiers, but there are models from many other manufacturers at a range of price points and power ratings. I've had personal experience with AER, Schertler, Fishman and Vox models, and have also heard good reports of Tanglewood acoustic amps, so all these makes are worth checking out.

Adding Up

Compact, active PA cabinets can be put to good use as extension speakers to complement acoustic amps or small PA systems.

Compact, active PA cabinets can be put to good use as extension speakers to complement acoustic amps or small PA systems.Compact, active PA cabinets can be put to good use as extension speakers to complement acoustic amps or small PA systems.



At larger venues, or those where a greater spread of sound is beneficial (those awkward L-shaped bars, for example), you can add an extension active speaker cabinet, as acoustic guitar amps usually include an XLR DI or link output, which can also be used to feed a large PA system at big gigs. A fairly small powered two-way cabinet (8- to 12-inch woofer plus horn) rated at anything from 100 Watts upwards will do the trick. I've used a Mackie SRM350, which has a 10-inch driver, on many occasions and found it to work well, and for smaller bar gigs I've even used a Mackie SRM150 powered mini-monitor as an extension speaker, with good results.

The Next Step



The human voice and the acoustic guitar don't usually produce low enough frequencies to require a great deal of bass extension, which is one reason why compact systems can produce great-sounding results. However, once you bring backing tracks, electronic keyboards or a full band with bass and drums into the equation, adequate bass handling becomes essential.



The solution you choose will depend on the range of venue sizes you tend to play. If you only play pubs, one of the smaller 'sub plus two satellites' systems may be the most appropriate, as these can still be compact enough to fit into a car, yet powerful enough to produce a satisfying musical experience. Good examples of this type of system include the smaller HK Lucas packages and LD Systems Dave setups. Because the sub takes over all the bass-handling duties, the satellite speakers can be made very compact — right down to shoe-box size — and even the sub itself might house only a single 10- or 12-inch speaker in the smaller systems. Most of these systems can produce a good stereo image even though there's only one sub, as very low frequencies are almost omnidirectional and it is difficult for the human hearing system to pinpoint the source.

For smaller venues, and with simpler 'singer/songwriter' type acts, a basic two-speaker system without subwoofers may be all that's needed.

For smaller venues, and with simpler 'singer/songwriter' type acts, a basic two-speaker system without subwoofers may be all that's needed.For smaller venues, and with simpler 'singer/songwriter' type acts, a basic two-speaker system without subwoofers may be all that's needed.Because it's difficult to combine small size with high efficiency, and because it requires more power to achieve the same SPL at low frequencies as it does at high frequencies, I'd consider a system power of around 300 Watts to be a minimum for performers using backing tracks, or a mainly acoustic band playing the pub circuit. In such situations, acoustic feedback often sets the operating level threshold well below the PA system's actual handling capacity. Just be aware that if there are bass instruments that need amplifying, you'll need a more powerful sub. See also the 'Line Arrays' box, as the smaller line-array systems lend themselves to anything from solo performers, duos and acoustic acts to rock bands playing pubs and smaller clubs — and most can be expanded for use in larger venues. I've used both the smaller Fohhn Linea 100 and HK Elements system with my own band, and we've always had a great vocal sound. Even using just a single 12-inch sub with the Linea, we've been able to put the guitars and keyboards through the system, but if you need more than a hint of drums and bass in the PA, moving to a 15-inch sub (or two) should be considered the minimum option.



If you're not ready to embrace the line-array approach (and a decent line array can still be quite expensive to a band on a budget), there's plenty of mileage left in conventional 12-inch-plus-horn active speakers. These can be used on their own where the primary purpose is to carry vocals, acoustic guitars and so on, or in conjunction with a sub where bass instruments and kick drums need a little help. In theory, many of these so-called full-range boxes boast a frequency response of down to 50Hz or so, but the bass sound from such a box is seldom good and bass frequencies eat up so much headroom that the overall system level will be severely compromised. In other words, if you need to amplify bass, add appropriate subs, don't thrash your main speakers. Both active and passive subs typically include crossovers that reduce the amount of low end reaching the main speakers, allowing the latter to give of their best in the vocal range and above.



Sub-and-top PA systems can be made extremely portable, and they have the advantage that you can take only the elements of the system that you need for each gig.Sub-and-top PA systems can be made extremely portable, and they have the advantage that you can take only the elements of the system that you need for each gig.While plastic speaker boxes, such as the popular Mackie SRM and JBL Eon models, are resilient and adequately loud, many of the cheaper (and some of the less cheap) plastic boxes can sound a little resonant and blurred in the lower mid-range when heard alongside a good plywood or MDF cabinet. Although wooden cabinets are often a little larger, their flat, straight sides can also make them easier to stack in a car. Whether you go for wood or plastic, however, one factor that is common to all of them is that the bigger the main driver, the more difficult it is for the designers to get the frequency ranges of the woofer and HF horn to meet properly in the middle. Ten-inch woofers have limited power handling but tend to team well with a horn, and the same is true of a well-designed 12-inch system. But once you get up to 15-inch, only the better designers seem to pull the trick off successfully, and the crossover frequency tends to fall right in the middle of the vocal range, so if it isn't done right you won't get a natural vocal sound.



Sub-and-top PA systems can be made extremely portable, and they have the advantage that you can take only the elements of the system that you need for each gig.A couple of decent 12-inch main speakers ('tops') teamed with one or two 15-inch or 18-inch subs will tackle just about any situation, from the local pub to a beer festival, and you have some limited scalability in that you can take just one sub for smaller gigs and both for larger ones, or in very small pubs you may not want to put the bass or kick drum through the PA at all, so you can leave the subs at home.



If you play a variety of venue sizes, it makes sense to buy a scalable system, where you can take only the components that are needed to do the job, rather than hauling the entire system around with you everywhere. For example, two smaller subwoofers may be more practical than one large one in this respect, as you can leave one or both behind when needs dictate. You can also add more top cabinets when you graduate to larger venues, although in my experience there are few venues requiring larger systems that don't have their own in-house PAs.



I've handled music days in the local park bandstand with everything from folk to punk using just a pair of Mackie SRM450s and a Mackie 15-inch sub, although in recent years I've been using HK Premium Pro 12-inch tops and a single 15-inch Fohhn sub. The Premium Pro cabinets fall at the budget end of the HK range, but they still produce very credible results if you don't need huge amounts of level. If you go further upmarket, to their Actor range, or use a premium brand such as Fohhn, you can cover a rugby pitch using as little as two 12-inch tops and two 18-inch subs (as I often do at Malvern's annual West Fest music festival).

False Economies



Where more volume is needed, especially if you need to put bass or drums through the PA, a subwoofer is pretty much essential.Where more volume is needed, especially if you need to put bass or drums through the PA, a subwoofer is pretty much essential.


Where more volume is needed, especially if you need to put bass or drums through the PA, a subwoofer is pretty much essential.
There are systems to cover every venue size and every budget but, as in most areas, you get what you pay for. In the world of PA, this tends to mean that spending more money not only gets you a better sound, it also gives you something more compact and portable. A high-end system might seem expensive, but it probably takes up less than half the space of a comparable system built using cheaper components, and most likely still sounds noticeably better. Spending a little more on the PA itself might mean that you can do away with the need for a van and take all your gear in cars, so factor these savings into the price.



Seemingly compact systems — such as the little line arrays I've mentioned — can, if well designed, fill surprisingly large venues, often carrying to the back of the room better than conventional box systems. They can be made so compact because all the heavy work is offloaded to the sub, leaving the relatively small mid-range/high-frequency units to carry the less burdensome part of the audio spectrum. Modern drivers are also more efficient than those of last generation, while Class-D amps pack an enormous amount of power into a much smaller and lighter package than is possible with a conventional analogue amplifier, so good doesn't have to mean heavy. A quality system will last a great many years, so it pays to choose carefully and to buy the very best you can afford.  

Line Arrays For Smaller Venues

Though they may seem expensive, modern compact line-array systems are very easy to transport and can offer extremely good coverage and sound quality.Though they may seem expensive, modern compact line-array systems are very easy to transport and can offer extremely good coverage and sound quality.Most of us have seen huge line-array speaker systems hanging above the stage at arena-sized gigs, often laid out in a vertical curve to provide even coverage from the front of the auditorium to the back. But in the last few years, the design ideas behind these large systems have been scaled down, and there's a lot to be said for using compact line arrays at smaller gigs, not least because of their small physical size when used in conjunction with a separate sub.

A line array is, at heart, a vertical stack of drive units, not unlike the modern equivalent of the column speakers used back in the '60s and '70s. The key benefit of a line array is that stacking drivers near-vertically changes the dispersion pattern of the resulting sound. Even a relatively modest line array measuring a metre or so long can exhibit a very tightly controlled vertical dispersion (as little as one-third of the horizontal dispersion angle). In practical terms, this means more sound is thrown onto the audience and less onto the ceiling or floor. Their horizontal dispersion tends to remain fairly wide too, so the overall coverage might be imagined as being fan-shaped.

A lesser known benefit of such a tightly-controlled dispersion pattern is that the sound level remains more consistent as you move further back in the venue than it does from a standard speaker box containing a woofer and a horn. Your audience will thank you for this, as you can work with a manageable SPL at the front of the room where the audience is close to the speakers, yet still have enough sound reaching the rear.

Good examples of compact line arrays include HK's Elements (10-inch subs plus columns of small-diameter mid/high drivers), the Fohhn Linea system (which uses a single separate tweeter, so bends the true line-array rules slightly), and LD's recent VA4 and VA8 setups. Fishman also make one without a sub for acoustic guitar players and singers.

The small size of such systems can lead to the incorrect impression that compact line arrays are too small for use with a conventional band, but my own experiences have confirmed that not only can they hold their own against traditional 'box' systems, they can actually sound more natural over the vocal range and produce more even coverage in difficult venues. My own view is that for pub and club gigs, the compact line array is the future, although they have a limitation of which you need to be aware: while conventional boxes can be stacked in a number of ways to increase the power handling, a line array can only be made longer (taller) — you can't put them side-by-side and still have them behave correctly. You can add more subs, of course, but the practical limit of, for example, the original HK Elements system, is around 8kW, using a line of 16 small drivers per side and eight 10-inch subs. (Just announced at the time of writing is a newer version that can accommodate up to 24 drivers per side in a stack of three eight-driver modules, in combination with their newly designed and more powerful 2 x 10-inch subs.)

Despite its appearance, the Bose L1 you may have heard about isn't designed as a traditional line array, as the drivers are angled to produce a very wide horizontal dispersion. I've attended practical tests against conventional line arrays set up to produce the same SPL at a given distance, where the Bose system sounded extremely good with very even coverage when the listener was reasonably close, but in a very large room the sound did drop away quite quickly with distance.

The Sub Way

You'll find a range of passive and active subwoofers on the market, and some of the better ones are capable of generating at least 125dB SPL. However, this measurement is only part of the story. Many subs are ported and tuned to sound impressively deep and powerful, but in reality their resonant tuning creates a degree of note 'overhang', making bass sounds less distinct than they should be. In other words, a kick drum that should go 'thud' ends up going 'boom'!
When you're auditioning subs indoors (such as in a shop or showrooom), it can be difficult to separate problems arising from the speaker design from those caused by the room, but at an outdoor event they could become very noticeable. Sealed-box subs create the tightest sound, but you need a fairly large box to allow a 15-inch or 18-inch driver to work efficiently at low frequencies. You can get away with using a pair of smaller subs with 10- or 12-inch drivers each, but the available SPL from a pair of smaller drivers is often somewhat lower than from one big one.

Korg Pa600 Video Manual -- Part 1: Introduction and Navigation

Korg Pa600 Video Manual -- Part 2: Sounds

Q. Are expensive multicore cables really worth the money?

I've been looking at multicores, and wondering how much quality matters. People say that only the best cables should be used, and I know a cheaper one would probably break sooner, but is there a massive difference between a $100 and a $300 one? Am I likely to suffer more crosstalk and interference with a cheaper multicore?




High-quality multicore cables can be very expensive, but whether or not you should spend the money may simply come down to how you plan on using them. A less expensive one may actually last very well if you're not planning on moving it from venue to venue, night after night.High-quality multicore cables can be very expensive, but whether or not you should spend the money may simply come down to how you plan on using them. A less expensive one may actually last very well if you're not planning on moving it from venue to venue, night after night.

High-quality multicore cables can be very expensive, but whether or not you should spend the money may simply come down to how you plan on using them. A less expensive one may actually last very well if you're not planning on moving it from venue to venue, night after night.

Via SOS web site



SOS Technical Editor Hugh Robjohns replies: A lot depends on how you plan to use the multicore. For example, if it's being gigged in different venues every night, and pulled and kicked about by roadies desperate to get a curry before the restaurant closes, a cheap one will probably lose channels almost daily and be completely dead within the month. On the other hand, if it's part of a permanent install in a private studio and used carefully only by you, a cheap one will probably last a lifetime.



The multicore cable itself isn't really the issue, although it is certainly true that better cables (by which I mean lighter, more flexible, with better screening, and designed to be easier to terminate) do cost more. However, most of the cost of a multicore system actually goes on the construction, and paying more generally means you get better quality connectors and a much better standard of internal wiring in the end boxes.



Systems with detachable end or breakout boxes are more expensive than those with permanently wired breakout boxes, because of the additional connectors and wiring involved. They are often easier to rig and de-rig, and can be more reliable in heavy-use situations, but the quality of the cable and box connectors is critical — and good ones are frighteningly expensive!    

Monday, January 19, 2015

Korg In The Studio - Krome Music Workstation -- TouchView Navigation Tips & Tricks

Q. How and when should I normalise my mix?

I have a question about normalising. I mix from Cockos Reaper through an M-Audio ProFire 2626 interface into an Allen & Heath ZED12 FX mixer, and then back into Reaper. I often find that the end mix level is lower than expected, and I have to push the master fader on the desk up over zero. I do set the gain correctly for each hardware channel, using the PFL button, and I have the outputs of the 2626 up at maximum. My worry is that if I start pushing up the levels of the individual track faders on the mixer, I'll start introducing unwanted hiss from the hardware, so I tend to mix to a maximum of unity gain on the individual channels — but maybe I should start going beyond that? Also, if I do normalise my mix, should I do it before I apply my master-bus processing (a bit of compression and limiting), or should I apply the master-bus processing first and then normalise that processed file to take full advantage of the available digital headroom?




Simple gain plug-ins, such as GVST's freeware GGain, are tremendously handy when mixing in a DAW, partly because they give you more control over the level arriving at any analogue-modelled plug-ins in your collection — a factor that can be crucial to their tone.Simple gain plug-ins, such as GVST's freeware GGain, are tremendously handy when mixing in a DAW, partly because they give you more control over the level arriving at any analogue-modelled plug-ins in your collection — a factor that can be crucial to their tone.



Timo Carlier via email



Simple gain plug-ins, such as GVST's freeware GGain, are tremendously handy when mixing in a DAW, partly because they give you more control over the level arriving at any analogue-modelled plug-ins in your collection — a factor that can be crucial to their tone.SOS contributor Mike Senior replies: The first main issue here is avoiding unwanted noise and distortion from your analogue components, and a good basic principle to bear in mind is to maximise your signal-to-noise ratio as early as possible for each piece of equipment, and then to leave any subsequent gain controls at their unity-gain position if possible. So in your situation, make sure the signals you're routing out of Reaper are making full use of the digital output headroom, so that you're sending the maximum level out of the ProFire 2626's sockets. (Check also that the faders in the ProFire's own mixer utility are set to unity gain, so that they don't undermine your efforts in Reaper.) Boosting the gain in your DAW at this point shouldn't incur any significant side-effects or additional background noise as long as you don't clip the output buses in Reaper.



With a good level coming out of the ProFire 2626, you may well find you need very little, if any, boost from the ZED12's channel Gain trim to give decent PFL readings, so from that point until you record the mixdown signal back into the computer, you should only need to turn things down. Clearly, this is what the faders are for, but you may also wish to use the channel Gain trims too, in some cases, in order to keep the channel faders for quieter sources closer to the unity-gain mark, where there's better control resolution. From that point, the trick is to build up your analogue mix so that it naturally fills the console's available output headroom. If you start with your first instruments too quiet, you'll end up with a low output level and therefore more background noise than necessary. If you start things too hot, you'll start clipping the mix bus before all the instruments have been added. It's a bit of a knack, so don't sweat it if you don't nail it exactly first time. If you under/overshoot, the best thing to do is adjust the channel faders en masse to redress the situation. Any channel insert processing will be left unchanged in that way, as well as any post-fader effects levels, so the amount of rebalancing you'll need to do is likely to be fairly small.



My main tip for getting it right first time is to fade up early any channels with strong transient peaks or powerful low-frequency energy: typically bass and drums in modern commercial productions. These elements take up the lion's share of the headroom in many mixes, so they provide a good early indication of the headroom the full mix is likely to demand. In fact, with a little practice, you should be able to discover 'rule of thumb' starting levels for your drums and bass on the ZED12's meters, which will usually lead to a good final mix level.



If you're doing your master-bus processing in the computer, just get the hottest clean output signal from the mixer into the audio interface. If the signal's too hot for the interface, turn down the mixer's master fader; if it's not hot enough, push up the console's master fader (if that doesn't clip the ZED12's output circuitry), or apply additional analogue input gain on the interface. Once the signal is digitised at a good level, it shouldn't need any normalisation before further bus processing; even 24dB of headroom should be perfectly fine at 24-bit resolution. If you're bus processing through analogue gear before digitising, just keep the same principles of gain-structuring at the front of your mind, and you shouldn't come too far unstuck: feed the processor as hot as possible without clipping it, and avoid adjusting the gain unnecessarily until you need to set an appropriate output level for the next piece of gear in the chain.



In addition to technical considerations, the level at which you drive any piece of analogue circuitry also affects the subjective sound in less tangible ways, and this concern may justify creatively modifying some of the above generalised tactics. If you hit the Allen & Heath's input circuitry a little harder than strictly recommended, for instance, you may find you like the resulting saturation harmonics. Perhaps your analogue bus compressor provides slightly different release or 'knee' characteristics (for the same gain reduction) if you process a high-level signal with a high threshold than if you work on a lower-level signal with a lower threshold. Or maybe the mixer's output bus might sound more transparent on acoustic music if fed more conservatively. Finding out that kind of stuff is one of the really fun bits about analogue mixing, so if you've already made the effort to get stuck into the hybrid analogue/digital approach, I imagine you don't need too much convincing to get your hands dirty there!



Finally, it's tempting to think that such signal-level concerns are pretty much redundant in the digital domain, but that's not really true, given how many emulated analogue plug-ins most people now seem to use. If this kind of processor is faithfully modelled, it will usually track the non-linear characteristics of its analogue forbear, so you still need to be aware of what level is hitting it. This is one of the reasons why my Mix Rescue projects are full of lots of instances of GVST's freeware GGain plug-in, a simple +/-12dB VST gain utility that gives me more control over the internal gain-structure of my channel plug-in chains. (There are also various built-in gain plug-ins within Reaper's Jesusonic set if you're not on a PC as I am.)