Welcome to No Limit Sound Productions. Where there are no limits! Enjoy your visit!
Welcome to No Limit Sound Productions
Company Founded | 2005 |
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Overview | Our services include Sound Engineering, Audio Post-Production, System Upgrades and Equipment Consulting. |
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Mission | Our mission is to provide excellent quality and service to our customers. We do customized service. |
Saturday, February 29, 2020
Friday, February 28, 2020
The Low-down On Analogue Interfacing
By Hugh Robjohns
With so much focus on computer workstations these days, wiring up analogue gear is becoming a bit of a lost art, but don't fear: all you need to know is here.
Despite the prevalence of digital audio in today's home studios, we still rely on analogue connections at some points in our signal chains. Analogue interfacing formats are stable and mature, but nevertheless it's surprising how often I am asked questions about connecting. This article is therefore intended to serve as a definitive guide to the basics.
Levels
The first thing to deal with is the topic of analogue signal levels. There are, fundamentally, five distinct nominal signal levels to be aware of. Starting at the lowest, we have 'microphone level' — the typical level you could expect to come from a typical microphone placed in a typical position in front of a typical instrument. Of course, there is no such thing: the actual level can vary wildly from microvolts (if you put a ribbon microphone at the back of a hall while someone plays quietly on a piano), to whole volts (from a capacitor mic inside a kick drum)! In general, though, most people would consider something like -50dBu to be a typical starting point for a nominal microphone level, and that equates to about 2.5mV rms. The 'rms' stands for 'root mean square,' a shorthand way of conveying the slightly involved method of describing an 'average signal voltage,' which is necessary because we are talking about varying AC signals here, rather than a steady-state DC voltage.
Hopefully, the use of the decibel is already familiar to most readers. The decibel is simply a logarithmic ratio of two signal amplitudes: one is the signal we are interested in, and the other is a stated reference value. The standard reference in professional audio is 0.775V rms, and we use the term dBu to denote it. The 'u' means 'unterminated' and refers to the absence of 600Ω terminations that used to be mandatory back when dinosaurs ruled the audio industry!
In broadcasting circles, the nominal 'line level' is called the standard 'alignment level,' and is defined as 0dBu, or a signal of 0.775V rms. However, most professional audio equipment actually employs a slightly higher nominal line-level of +4dBu, or 1.223V rms. You will often see and hear references to professional equipment as having "+4" interfaces.
Semi-professional and domestic equipment is generally designed to use a lower standard line-level, which is described as "-10." However, there is a trap here: a different level reference is used — this is -10dBV. The 'V' denotes a reference of one volt instead of 0.775V, and that means that the semi-pro line level of -10dBV equates to 0.316V.
The important practical point of this is that a -10dBV signal is almost exactly 12dB lower than the +4dBu professional line level. Which explains why, if you connect a semi-pro device to a professional one, the average signal level is about 12dB quieter than expected and, going the other way, a professional signal peaks roughly 12dB higher than a semi-pro device expects!
Another important signal level to be aware of is the maximum 'headroom'. Well designed professional equipment will run out of headroom and start to clip at about +28dBu, which means a whopping signal level of about 20V rms — although I've seen some really impressive equipment that can cope with more than +36dBu! Less well designed audio devices usually have lower headroom margins. For example, +22dB is fairly typical, and semi-pro devices will have even lower margins because of their lower nominal line level.
Balanced Or Unbalanced?
The next key concept is that of the balanced interface, which is probably the most misunderstood concept in the entire world of audio! Most people know that a balanced interface involves three wires: hot and cold signal wires, plus a screen connection; and an unbalanced interface has just two: signal and screen/ground. Most people also know that whereas the screen/ground is essential in an unbalanced connection because it provides the signal return path, it plays no part in the signal path for a balanced interface, and is purely a screen to reduce interference. That's why you can remove the screen connection of a balanced interface to prevent a ground loop, without losing the signal.
There is an element of truth in there somewhere, but it is not the whole story and it is rather misleading.
In fact, the operation of the balanced line relates directly to the Wheatsone Bridge... I expect you wished you had paid more attention in school physics lessons now! The critical aspect of any balanced interface is that its design forces any interfering signals to have as near as possible the same amplitude on both wires. That is the essential aspect that allows the differential input to cancel out the interference — and that condition is ensured by arranging for each signal wire to have an equal impedance to ground.
So it doesn't actually matter to the operation of the balanced interface whether there is a pair of symmetrical and inverted signals or not. The point is that given equal impedances to ground from each signal wire, any interference will be presented to the differential receiver as a 'common mode' signal, which means equal in polarity and amplitude on the two signal wires. The differential receiver (whether a passive transformer or an active electronic input) inherently rejects common-mode signals — removing the interference — while passing any differential signal, which should be the wanted audio.
This kind of configuration is often referred to as an 'impedance-balanced' or 'ground compensated output' and, typically, the hot side of the balanced interface is driven to carry the required output signal, while the cold side is connected to ground at the source end via a resistor which is carefully chosen to make sure that both wires have the same impedance to ground. (A true 'ground-compensated' output also monitors the voltage on the cold side and feeds that into the driving amp to cancel out any ground-loop noise.) I know many people look at impedance-balanced and ground-compensated outputs with scepticism, and don't believe they are a true balanced output at all — but they really are perfectly functional balanced interfaces, with exactly the same interference-rejecting properties as the more familiar, symmetrically driven system.
The only practical difference is in terms of the signal amplitude on the wires. The hot side of an impedance-balanced or ground-compensated output typically carries a signal which is 6dB larger than the level carried on each wire of a fully symmetrical balanced output. If you think about it, it has to be that way to ensure compatible output levels from a differential line receiver. The fact that only one line is driven in a ground-compensated output is also advantageous in situations where the output is likely to be connected to an unbalanced input, since the signal level remains as expected. If you connect a symmetrical output to an unbalanced input you normally lose the cold side contribution, and the signal level ends up 6dB lower than it should be!
Patchbay Wiring
Once the heart of any studio, patchbays, or jackfields, aren't as popular or necessary in computer-based facilities. However, where you need to be able to patch analogue outboard equipment into different signal paths, patchbays remain the most cost-effective solution.
Most commercial home studio patch-panels are designed for A-gauge plugs (see main text) and have some form of user-adjustable 'normalling', either via switches or by removing and re-orientating PCB cards holding the socketry. The normalling in professional B-gauge and Bantam jackfields is configured by wiring the appropriate terminals and switch contacts as necessary on each socket — not a job for a soldering novice!
Configurations vary, but generally patchbays are configured with two rows of jack sockets per rack strip. The upper row (as seen from the front) is normally used for outputs, while the lower row is used for inputs. For example, the upper row might carry the outputs from a multi-channel recorder, while the lower row provides the channel inputs to a mixing console.
To make a connection, you simply insert a suitable patch cord to link the appropriate output to the required input destination — giving each plug a couple of twists as you insert them to make sure the contacts are wiped clean.
In most cases, there will be a standard default set of connections, such as the recorder outputs to the desk channel inputs. It is obviously tedious to have to plug patch cords for this kind of thing, so it is common practice to 'normal' the connections instead. This means arranging for the output signal on the top row of jacks to be wired inside the patchbay to a set of break contacts on the corresponding input sockets directly below.
In this way, with no patch cords plugged in, the recorder outputs are passed automatically to the corresponding desk channel inputs. This is called a 'half-normalling' arrangement. If you need a second output from a recorder channel, you can simply plug a patch cord into the appropriate output socket on the patchbay without breaking the signal to the desk channel.
However, if you need to route something else into a particular desk channel, instead of the normalled recorder output, you can simply plug a patch cord into the required channel input socket. Inserting a plug forces open the break contacts, interrupting the signal flow from the normalled output jack above it. For this reason the input socket is sometimes called a 'break jack.' The diagram shows the wiring arrangement, and exactly the same configuration is used for A- and B-gauge and Bantam jackfields.
There are a couple of alternative forms of normalling, although these are less commonly used and rarely available on commercial patchbay panels. Single Normalling is where the breaking contacts of both the output and input socket are wired to each other, so that inserting a single plug into either the output or input socket breaks the normalled link. Double Normalling is effectively doubled-up half normalling, and to break the normalled signal path it is necessary to insert plugs into both the output and input sockets. I can't think of an installation in which I've used single normalling, but double normalling is often used for main outputs and other mission-critical connections. The advantage over half or single normalling is that there are two sets of break contacts wired in parallel to pass the normalled signal, which improves reliability significantly.
Obviously, you can't see the normalling arrangement behind the sockets of a patchbay, so to indicate which sockets are normalled it is usual to colour code the labels above each socket. The BBC convention is to show half normalling with red above the input (break) sockets, while double normalling is shown with yellow above both output and input socket — but this is only one convention and different manufacturers and organisations use alternative colour schemes.
When designing a new patchbay layout, it is important to plan the location of each group of signals carefully and to make sure the normalling is logical. The classic mistake is to have the output of an effects unit normalled to its own input, which isn't a good idea! Usual practice for a small installation is to have the signal flow zig-zag across the patchbay from left to right: source outputs to channel inputs, group outputs to recorder inputs, recorder outputs to monitor inputs, desk outputs to master recorder inputs, speaker outputs to amplifier inputs... and so on.
Connectors
There are five commonly used general-purpose analogue audio connectors: the XLR, the B-gauge jack plug, the Bantam/TT plug, the A-gauge jack plug, and the RCA phono plug.
The three-pin XLR is probably the connector most strongly associated with professional audio equipment, used for both mic and line-level interfaces. The name actually refers to the original Cannon X-series connector, which was then upgraded with a Latch and a Rubber insulator — hence the XLR. Although now made by many alternative suppliers, the XLR-format connector has been the standard for most microphones since the late 1970s, and the AES organisation eventually managed to standardise the wiring convention worldwide in 1992, in a document called AES14.
Balanced, stereo microphone and line-level signals can also be conveyed using a five-pin version of the XLR. Again, the screen is on pin 1, with the left channel on 2/3, and the right channel on 4/5. The even pin-numbers carry the hot side of the balanced interface and the odd numbers carry the cold side. The same connector convention is used, with female for inputs and male for outputs.
XLRs are very rugged connectors, with self-cleaning and gas-tight contacts in the best designs, and provide excellent screening properties. However, they are quite bulky and impractical where many connectors have to be squeezed into a small space. The original solution to this problem was the Post Office 316 plug, known generically as the B-gauge plug. Developed for manual telephone switchboards, this brass quarter-inch plug has three contacts arranged as a rounded tip, a ring and a longer sleeve. The tip normally carries the hot side of a balanced signal, the ring carries the cold side, and the sleeve carries the screen.
For even greater socket density — such as might be required on built-in console patchbays — there is a miniature version of the PO316, called the Bantam or TT (Tiny Telephone) jack plug. Essentially, the Bantam is a scaled-down version of the PO316 with a 4.4mm diameter instead of the latter's 6.35mm diameter. That allows 48 sockets per row, doubling the socket density compared with a standard PO316 patchbay. Largely because of the smaller size, Bantam patchbays don't tend to be quite as reliable as the full size B-gauge version, but in most cases this shortcoming is outweighed by the space-saving convenience. Bantam plugs are usually employed for line levels at +4dBu.
In the semi-professional world, the A-gauge jack plug is often used in an equivalent way to the B-gauge for balanced circuits. Again, the TRS (tip, ring, sleeve) version carries hot, cold and screen connections, respectively. When used for balanced line-level signals, A-gauge plugs and sockets could work at either +4dBu or -10dBV, depending on the intended market for the equipment. Some equipment even allows the level to be switched to suit the application.
A-gauge jacks are also used for unbalanced circuits, but in a simpler TS (tip-sleeve) version, typically used for guitars and keyboards. The TRS version is also used for unbalanced stereo connections, most commonly on headphones. In this case, the wiring convention is left channel to tip, right channel to ring, and the commoned returns to the sleeve — and that applies to the miniature 3.5mm TRS plugs on stereo earpieces and headphones too.
Although the A-gauge and B-gauge plugs have the same overall shaft diameter, they have completely different tip shapes and the dimensions and positions of the insulators between conductors is also different, which makes them completely incompatible with each other. The most significant difference is that the B-gauge has a small, rounded tip, while the A-gauge has a larger, diamond-shaped tip. If a B-gauge plug is inserted in an A-gauge socket, it is unlikely to make reliable contact because of the smaller tip, whereas if an A-gauge plug is inserted into a B-gauge socket its oversized tip is likely to strain and bend the contacts, such that they then become unreliable with proper B-gauge plugs. So the golden rule is to never insert the incorrect type of plug for the socket: a handy rule of thumb for life, I find!
The outer ring of the RCA phono plug and socket are used for the screen connection, while the central pin/socket is used for the hot signal connection. The spring action of the outer contact is all that holds the RCA connector in place, and cheap phono plugs have a nasty habit of losing their grip over time. This makes the screen connection unreliable, and sometimes the plugs can even fall out of the socket!
I've only described the five most common connectors here, but there are many other types used for specific applications. For example, vintage mics and some older audio equipment use variations on the multi-pin DIN connector (Tuchel, Preh and Binder), while loudspeakers are usually connected via 4mm banana plugs, or Speakon connectors. Multicore cables are typically equipped with Plusconnection, Contact or MIL connectors, and multi-channel interfaces to tape recorders and DAWs are often via D-sub (see photo above) or EDAC connectors. There are also numerous variations on wiring standards and pin-outs, just to add to the confusion. Hopefully, though, this article will have clarified the most important aspects of analogue interconnection. Happy plugging!
Published May 2007
Thursday, February 27, 2020
Wednesday, February 26, 2020
Q. Should mixer insert connections be balanced or unbalanced?
By Hugh Robjohns & Paul White
I’ve read that pro gear should use balanced connections, but I’ve been looking for a mixer recently and find that most have unbalanced insert points. My interface uses balanced inputs and outputs, as does most of my (modest collection of) outboard gear. What gives?
Bruce Milner via email
SOS Editor In Chief Paul White replies: Many mixers offer unbalanced insert points as it’s convenient to use a TRS jack to handle both the send and return signal. It also saves on balancing and unbalancing circuitry inside the mixer. In most cases, the unbalanced cable runs connecting the external gear are short enough that interference won’t be a problem, especially as the signals are at line level, although there’s the possibility of increased hum due to ground loops. This issue is caused by having multiple ground paths between equipment, due to having both mains power grounds, and the unbalanced cable screens on connecting cables also being ground conductors.
In reality, this is not usually a significant problem, though if you were to actually measure the hum level using sensitive test equipment, it is likely that it would be higher in an unbalanced system even if still inaudible. In the event that ground-loop hum is evident, some improvement can be achieved by making up special cables to connect the insert points to your patchbay or external equipment.
SOS Technical Editor Hugh Robjohns adds: Paul mentions making some ‘special cables’. It’s worth mentioning that this sort of DIY job needn’t be daunting! With that in mind, here’s a diagram (above) which should help anyone wanting to adapt a TRS to 2xTS or 2xXLR insert cable to allow balanced gear to be used with an unbalanced single-socket insert point.
Published December 2015
Tuesday, February 25, 2020
Monday, February 24, 2020
Q. Can I lift the ground on a MIDI cable?
By Hugh Robjohns
Hi there, I’m using some old-style MIDI cables to connect some older synths, however I’m finding that when I plug in the MIDI cables it causes a hum through my speakers, which I suspect is due to a ground loop. I know you can sometimes disconnect the screen wire on certain cables and this eliminates the problem, is it possible to do this on a MIDI cable?
SOS Forum post
SOS Technical Editor Hugh Robjohns replies: The short answer is, yes, you can — but given that your MIDI cables may be revealing a pre-existing ground loop rather than causing a new one, I’d start by taking all the standard precautions to avoid ground loops.
First, make sure that all your connected audio equipment is powered from a single mains wall socket in a sensible way — which means using just one single or double wall outlet and distributing the mains to the equipment via a star-arrangement of plug-boards. Next, consider the audio connections and think about using line-isolating transformers or pseudo-balanced cabling if you need to connect unbalanced equipment with Class-1 (grounded) mains power supplies to other devices. (Any double-insulated or Class-2 mains-powered equipment doesn’t have a mains safety earth connection, and so is unlikely to cause or suffer from ground loops.)
If all of the mains and audio wiring is up to scratch and ground-loop free, then examine the wiring of your specific MIDI cables. Although the MIDI input device uses an opto-receiver to isolate the two data lines (connected between pins 4 and 5 of the DIN plugs), the cable is normally screened and that screen is typically connected to the equipment grounds at both ends via pin 2, which is where the potential for a new ground loop occurs.
Sometimes a sneaky ground loop is formed because the DIN plug’s metal body is connected to the cable screen inside the plugs, so check for that and remove those connections at both ends, if present. If you still have a problem you may finally have to resort to cutting and isolating the cable screen’s connection to pin 2 (the middle pin), but ideally only at the MIDI in end. However, be aware that this solution can disable the power supply return path if the MIDI receiving device also expects to be powered over its MIDI connection — that’s unlikely, but still something to bear in mind.
Published May 2016
Saturday, February 22, 2020
Friday, February 21, 2020
Q. Is it safe to run everything off one power outlet?
By Hugh Robjohns
I'm about to move house and set up a studio in the spare bedroom with all my stuff, and I'm just interested in what the best use of power points is. Is it acceptable (and safe!) to run many plug boards (between 24 and 28 at least) from one outlet in the wall, or is it better to use all the available outlets around the room? I have quite a lot of stuff to power in one small room, and I am thinking about perhaps getting a power conditioner as well — I don't want to be responsible for burning down our landlord's house!
SOS Forum Post
Technical Editor Hugh Robjohns replies: From the point of view of avoiding ground loops, it is best to run everything from a single socket, or from adjacent sockets if you have a double-socket outlet. This is a much better approach than running some gear from a socket on one side of the room, and other gear from a socket on the other side — a practice almost guaranteed to produce ground-loop problems!
If you are concerned about the total power you will be drawing from a single socket, you can reduce the load by plugging non-audio equipment into another socket in the room — desk lamps, phone chargers, kettles and so on.
If you're using plug boards to increase the number of available sockets, connect them in a star arrangement rather than serially. By that I mean you should connect one board to the wall socket, then plug the other boards into that first one, and then plug the equipment into this second 'layer' of boards. That way, the earth paths are kept as short as possible and in a star arrangement.
A single socket is able to supply around 3kW (230V x 13 Amps) and it is very unlikely that your domestic recording equipment will draw that much power — but every piece of equipment will have a label on it near the power connection that says what power (or current) it draws. If possible, try to balance the power demands on each plug board and make sure that all the fuses (in the plug boards and in individual plugs) are sensibly rated. Also, bear in mind that the first plug board has to carry the entire current load.
Published October 2005
Thursday, February 20, 2020
Wednesday, February 19, 2020
Q. How can I permanently stop mains noise in my studio?
By Martin Walker
Can you recommend products suitable for the European power grid that can be used to clean up the power signal and ground loops? I am experiencing both ground loops and a generally dodgy power signal. A lot of people recommend that I use some sort of UPS (Uninterruptible Power Supply), but I don't need the functionality they provide, and I would rather spend money on better power conditioning and filtering equipment. Your advice will be greatly appreciated!
Alexander van Rijn
PC music specialist Martin Walker replies: In my opinion it's only worth 'cleaning up the power signal' if it's dirty, and a huge number of background noise problems are caused not by mucky mains, but by audio wiring that results in ground loops. This is the source of lots of unwanted nasties that sneak into your audio signals, and removing them often requires no dedicated products at all. Problems range from straightforward 'hums' (which normally include various levels of the mains harmonics, such as 50Hz, 100Hz, 150Hz, and so on in the UK, or 60Hz and higher multiples in the US), to a wide range of scratches, ticking, buzzing and other digital gremlins that are often associated with computer activities such as graphic redraws, mouse movements, and hard-drive activity.
If you're experiencing any of these ground-loop problems, you won't solve them by installing a power conditioner or an Uninterruptible Power Supply, so before you even think of spending money on either of these options you should examine your basic wiring. Temporarily unplug all the audio cables from your setup, and if you've got gear bolted into a rack, it may also be worth disconnecting the mains cables of this other gear to rule out problems with several metal cases touching each other and causing yet more ground loops.
As tempting as it might seem, short cuts such as leaving the cables plugged in and just switching off the connected gear at the mains won't work, since the mains cables and any resulting ground loops will still be in place. Unplugging one cable can therefore make the background noises better or worse, depending on how this affects the remaining ground loops. Only by removing every audio cable and working through your studio item by item can you totally eradicate ground-loop problems.
You should now hopefully hear silence from your loudspeakers or headphones, apart from a little hiss and possibly a tiny amount of hum or buzz if you turn the amplifier right up and place your ears nearby (be very careful when doing this, since an unexpected signal at this point could damage your ears or blow up your speakers). If there's still more hum than you expect, it might be due to a nearby 'line-lump' power supply, in which case, you should move this as far as possible from audio cables, and at the very least try rotating it to find the 'quietest' position. If you're still unhappy with the levels of hum and noise from your amp/speakers you may need to get them checked out by a technician — remember that hum levels of both solid-state and valve amps can increase over time, due to deteriorating capacitors or valves.
Assuming all is well at this stage, turn down the speaker levels, connect your mixer to the amp, turn up and listen again (if you route all your gear directly to a multi-channel audio interface, this is your 'mixer'). You'll probably hear greater hiss levels from the combined contribution of all the input channels until you pull the master fader right down, but there still shouldn't be any obvious hum or other interference. If there is, it's generally because you've just created an earth loop — the amp/speakers are already earthed via their mains cable, and the mixer is earthed in exactly the same way, so when you connect the two with an audio cable its screen connection completes the loop, causing unwanted earth currents to flow.
If your amp has balanced inputs and your mixer/interface has balanced outputs, the cure is to connect the two via balanced audio cables (twin core plus screen). If not, you may be able to achieve the same results by disconnecting the screen of an unbalanced cable at one end (in the case of soldered cables you can do this inside the plug, normally at the destination end). Similarly, if the amp has a balanced input, but your mixer/interface only provides an unbalanced output, you can make up a pseudo-balanced cable, as I described in 'Computer Audio Problems' in SOS November 2004. Here, one end of the balanced cable is wired to a balanced jack or XLR as normal, while the other end is wired to an unbalanced jack with the screen disconnected or, preferably, connected via a resistor. These cost only a few pence more to make than unbalanced cables, yet provide an ideal solution for connecting any unbalanced source to a balanced destination. I've got such cables wired between all my hardware synths and mixer, and background noise levels are considerably reduced as a result.
Occasionally the only way to cure a ground-loop problem is to install a line-level DI (Direct Injection) box between the mixer and amp, to 'galvanically separate' the two circuits, commonly by using a transformer to transfer the signal — the audio gets through perfectly, but there's no direct connection at all between the input and output cables inside the DI box. This is sometimes the only way to cure some laptop-related ground-loop problems, but in my experience, most others can be dealt with by cable modifications.
Once your mixer, amp, and speaker chain have an acceptably low level of background noise, plug each remaining item of gear into your mixer in turn and power it up, listening at each stage for unwanted noises. As soon as you hear any, you know you've either got a faulty piece of gear or a ground-loop problem, and can sort it out in exactly the same way as before. If it's rack gear, you may need to temporarily unbolt it from the rack to check that the problem isn't due to its case touching other earthed metalwork and creating a further ground loop (if it is, use nylon rack bolts or 'Humfrees' to isolate it).
Alternatively, low-level circuitry such as mic preamps can pick up mains interference from the mains transformer inside a nearby rack unit. This systematic approach is the only way to deal with ground-loop problems. It may be tedious, but you only have to do it once, and the benefits can be enormous!
When you've got all your gear connected, and still have no hums or other nasties, then and only then is the time to consider adding a 'power conditioner' or UPS. A power conditioner will filter the mains signal to remove any radio-frequency interference plus any incoming spikes and other intermittent noises riding piggyback on the mains signal from the outside world. However, most modern electronic gear, including computers, already includes such filtering in its own power supplies, and in general, it's far better to suppress switch-related mains transients from distant devices such as refrigerators and central heating systems at source, as this will be far more effective.
If, after solving your ground-loop problems, you don't hear any other nasties then you probably don't need a power conditioner at all, but they can be very useful bolted into a rack for live use, to cope with unexpected 'incoming' problems due to stage lighting or grotty wiring in unfamiliar venues. However, if your mains power is 'generally dodgy' it may pay you to have an electrician check your house wiring, and contact the local electricity board to have your incoming mains checked for quality. If, for instance, you live in a remote rural location or close to an industrial estate, you may suffer from occasional but unavoidable interference problems that will benefit from a studio-based power conditioner, although I've never personally found the need for one (perhaps I've been lucky).
A UPS will, in addition, cope with 'brownouts' (occasional severe drop in mains voltage, generally for a few seconds only), plus the more severe 'blackouts' (complete loss of mains power), in exactly the same way as a laptop computer carries on running on battery power if you pull its mains plug. Even if you only use the UPS to power your desktop computer rather than the whole studio (generally a far cheaper approach), it can prove invaluable if you have paying clients in your studio, to avoid your computer rebooting in the middle of a session, and can give you a vital few extra minutes to save the current project before the UPS backup power runs out.
Published July 2005
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