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Frequently Asked   Questions

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Please note that these FAQs are provided to suggest solutions to technical problems. If your problem is not covered here then please feel free to contact PAforMusic.

"The only thing more expensive than education is ignorance." − Benjamin Franklin.

1. Which speakers / microphones / mixer / amplifiers / etc should I buy?  View
2. How can I avoid feedback?  View
3. How can I solve a hum (or buzz) problem?  View
4. How can I avoid background hiss?  View
5. How powerful does my system need to be?  View
6. How should I set up my channel Gain controls?  View
7. How should I set up my channel EQ controls?  View
8. Should I use mono or stereo?  View
9. How should I mike up a ...?  View
10. How do I wire up a microphone cable?  View
11. My amplifier has a quoted output power into 4 ohms, but my speakers are 8 ohms. Is this a problem and how much output power will I get?  View
12. Why do my speakers have different quoted maximum power values for 'RMS' and 'music power'?  View
13. I changed my speakers − why do they sound louder (or quieter) than my old ones, even though they have the same impedance value?  View
14. How do I connect the balanced output from my mixer to the unbalanced input of my power amp, without getting hum problems?  View
15. How do I calculate ...?  View

Q 1.
Which speakers / microphones / mixer / amplifiers / etc should I buy?

A 1.
PAforMusic cannot provide answers to questions such as "Which speakers should I buy?" because choosing the most suitable equipment (of any type) depends on so many factors. Often compromises between the different factors may be necessary, depending on their relative importance in your particular case. Some of the most important factors will usually include:

• How large is your budget?
• How large are your venue(s) / audience(s)?
• Indoor or outdoor? (e.g. weather-resistant speakers)
• How large is your band? (or how many simultaneous performers of whatever type?)
• What maximum sound level (SPL) is required, and what variation in SPL throughout the venue is acceptable?
• What kind of music will the equipment be used for (or just for speech?)
• How important to you (or to your audience) is sound quality?
• Is ease of transportation and/or assembly important? (compact size? light-weight? rugged construction?)
• How important to you is reliability? ('build quality')
• Are visual aesthetics important? (e.g. speakers in architecturally sensitive venues)
• How might each of the relevant factors change in the near/mid future?

Information on specific items of equipment can be downloaded from many of the manufacturers' websites − see the Equipment Manufacturers page. Some UK suppliers are listed on the Suppliers page.

Above all, be sure to select equipment that fully meets your most important requirements − regardless of what attractive 'extra' features it offers.

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Q 2.
How can I avoid feedback?

A 2.
Acoustic feedback, the most common variety, results from too high an overall gain around a complete loop through the PA system, from microphone, through mixer and amplifier, to speaker and back (through the air) to microphone again. To avoid the feedback, the overall gain around this loop at the feedback frequency must be reduced. This may be achieved through one or more of the following actions (not all of which will be relevant or practicable in every situation):

• Ensure that every microphone is situated as close as possible to the sound source that it is intended to pick up, taking into account the directional and tonal characteristics of the sound source, the maximum sound level that the microphone can operate at and the proximity effect of the microphone.
This will produce the highest possible output level from the microphone(s) and so require the least amount of gain from the PA.
• Ensure that only the microphones that are really needed are open (active), at any point in time. The 'Mute' button on the mixer is useful for cutting the signal from a mic without losing your fader setting. If you don't have these buttons, turn the faders of unused microphones down.
• Keep monitor speaker sound levels down to the minimum that is really needed by the performers.
• Always observe the basic positioning order: Performer, microphone, front-of-house (FOH) speakers, audience. (So, when you stand directly in front of the front-of-house speakers, with your back to them, you shouldn't be able to see any microphones. Likewise, when you stand at the location of any microphone and look around, you shouldn't be able to see the front of any FOH speakers.)
• As far as practicable, the microphones should point in the opposite direction to the FOH speakers.
• Discourage users of radio-microphones from walking in front of the FOH speakers.
• Position the monitor speakers behind the microphones (that is, on the opposite side of the stand to the user), and angle monitors so that so that they point at the microphone's point of minimum pick-up. For cardioids, the monitor should be on the rear axis of the mic, i.e. directly behind it. For super-cardioids use two monitors, each at an angle of 55 degrees from the rear axis. For hyper-cardioids the two monitors should each be at an angle of 70 degrees from the rear axis.
• Encourage users of hand-held microphones in correct microphone technique. Microphones should be pointed directly at the user's mouth, at an appropriate distance for the type of microphone, loudness of the person's voice, and the amount of gain required from the system. (An appropriate distance is usually between 1 and 6 inches, or 2.5 to 15 cm − see proximity effect and Microphone Technique on the Getting Started − for Performers page).
• Discourage users of hand-held microphones from wrapping their fingers around the basket of the microphone.
• Discourage users of hand-held microphones from pointing their microphones at any other sources of sound, such as monitor speakers, combos, a drum kit, etc., or at any reflective surface such as a hard wall, ceiling or floor.
• Use good quality uni-directional microphones with an appropriate polar response pattern. (If in doubt, cardioid is usually a safe choice).
• Use good quality speakers (this applies to monitors as well as to FOH).
• Ensure that FOH speakers are correctly located and directed, and that they have appropriate directivity patterns to cover just the required audience area. Avoid directing them towards reflective surfaces such as hard walls, ceilings or floors.
• If possible, use in-ear monitoring (IEM) instead of monitor speakers.
• Ensure that the mixer equalisers on microphone channels are set appropriately (beware large amounts of boost).
• Use a graphic equaliser on the monitor mix, to reduce the system gain at the problem frequencies (see Ringing out) − being careful to avoid destroying the clarity of the sound for the performers. (Note that graphic equalisers, when incorrectly set, can also be the cause of feedback − beware large amounts of boost over narrow ranges of frequency.) Some types of graphic equaliser incorporate circuitry to detect and automatically eliminate feedback, but the effectiveness of such devices depends to a large extent on the particular circumstances.
• For a theoretical design approach to feedback control, see Potential acoustic gain.

If your system has an induction loop, and if the feedback is more of a "screech" than a whistle or a low note, and starts and stops suddenly, then it is possible that the feedback is magnetic rather than acoustic − you can check by seeing if the problem goes away when you switch off the induction loop. This kind of feedback occurs when the magnetic field of the induction loop is picked up too much by other parts of the system − most usually by electric guitar pick-ups. It can usually be avoided by changing the location or angle of the guitar, or by adjusting the mixer equalisers for the guitar channel. Preferably, use guitars with humbucking pick-ups, and route the loop cable so as to avoid including the stage area within the loop.

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Q 3.
How can I solve a hum (or buzz) problem?

A 3.
This depends on where the hum is being introduced into the system, and on the way that it is being introduced.

Considering the "where" aspect of this first, you should determine if the hum is coming into the mixer on one (or more) of the channels, and if so which, by the following procedure:

• If turning all the master faders (including all Aux Send masters) down to minimum does not eliminate the hum, then the problem is most likely with the amplifier(s), with the mixer-to-amplifier interconnections, with any equipment wired between the mixer and the amplifiers (such as graphic equalisers or active crossovers), or with the mixer itself.
• If the above action does eliminate the hum, restore the settings of the master faders and disconnect all the mixer inputs (including any local sources). If the hum still remains then disconnect any outboard effects units etc.. If the hum still remains then the problem is most likely with the mixer.
• If the hum disappears when all inputs are disconnected, then the hum is probably coming into the mixer on one (or more) of the channels − reconnect each in turn to find out which one. (If the hum appears to be on "every" channel, then there may be an earth loop problem − see below.)

Now considering the "way" aspect, there are five main ways that hum can be introduced:

1. By inductive coupling from a source of a mains-frequency magnetic field − usually from a mains transformer within an item of equipment, or occasionally from mains cables. Possible solutions in this case are:
   • Increase the distance between the equipment or cable which is producing the offending field and the equipment or cable which is picking it up.
   • Use a system with balanced inputs to the mixer, and ensure that all connections to balanced circuits are made using only balanced cables.
2. By capacitive coupling from mains cabling (not necessarily ones that are powering equipment associated with the PA system!). Possible solutions in this case are:
   • Check that screened cable is used for all connections except for speaker and mains connections.
   • Check that the cables or their connectors are not faulty (e.g. a screen not properly connected).
3. By means of an earth loop. This is a condition whereby a "circular" arrangement of earth connections exists because of the interconnection of two or more items of earthed equipment (typically backline amplifiers and mixers). Earth leakage currents flowing in the earth connections develop a hum voltage between the chassis of the interconnected equipment, which effectively adds to the signal voltage passed between them. To avoid this problem:
   • Make use of balanced feeds wherever possible. This enables the destination of the signal to largely cancel out the hum that it receives.
   • For the connection of backline amplifiers and other instruments with an unbalanced output, use a DI box, as close to the signal source as possible, to balance the signal. These often also have the facility to provide electrical isolation of signal earth connections, and so reduce hum even further. (Note that if phantom power is in use then a DI box must always be used.)
   • Never attempt to avoid earth loop problems by the disconnection of safety earths, as this would create a lethal hazard.
4. Because faulty or poor-quality equipment is directly contributing the hum into the signal passing though it. Or, because it has a 'pin 1 problem'.
   • Get it fixed or buy better equipment!
5. Because of a problem with the mains power arrangements.
   • Get it investigated and fixed by a qualified expert − and urgently as there may be an associated safety hazard.

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Q 4.
How can I avoid background hiss?

A 4.
The background "hissing" noise introduced by equipment can never be totally eliminated, but it can usually be rendered insignificant in relation to the level of the signal by ensuring that the equipment is carrying a level of signal that is no lower than the minimum that the equipment is intended to handle.

In practice, this usually means ensuring that the original source signal (e.g. from a microphone or instrument pick-up) is initially adequately amplified by a sufficiently high-quality pre-amp, and is not subsequently unduly attenuated (e.g. by a level control being turned down too much). For guidance on the correct setting of pre-amp gain controls see the Setting gain controls item on this page. For further information on minimising hiss levels, see Gain structure and Signal-to-noise ratio.

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Q 5.
How powerful does my system need to be?

A 5.
You first need to know how loud you want your system to be. This will depend upon the type of programme material, the size of your audience, the level of other sound sources (e.g. audience sound and background noise), etc.

As the sound level will decrease with increasing distance from the speakers, you need to consider the average sound level (SPL) that is needed at the front and at the back of the audience area. Then you can work out the sound level required at a distance of 1 metre in front of the speakers (see Inverse square law).

The amount of average electrical power (watts) that is needed to create a particular average sound level 1 metre in front of the speakers depends on the sensitivity of your speakers. More sensitive speakers give the same sound level output for less electrical power input. For example, a speaker with a sensitivity figure of 90 dB/W will need a power input of 1 kW to provide 120 dB SPL at a distance of 1 metre, but a 96 dB/W speaker only needs an input of 250 W to give the same sound level at that distance.

Your speakers will need to have a continuous average power rating that is at least as high as the average power input level that they require to produce the average sound level that you want. For example, if the sensitivity value of your speakers means that they need 500 W in order to give the average sound level that you want, then they will need to have a continuous average power rating of at least 500 W if they are to produce that sound level without being damaged. (The maximum continuous average power input level that a speaker can handle is often referred to as its RMS power rating, though this is strictly incorrect terminology.) For more information see Speaker Sensitivity on the Amplifiers and Speakers page.

Next you need to consider how much headroom you require − this will depend on how 'peaky' your programme material is, and on whether you are using a limiter. The important thing to ensure here is that the amplifier is able to deliver the peak power that is required without clipping (preferably with some margin to spare). For example, if the average power level is 500 W per channel and the amplifier has a power rating of 2 kW per channel then this will provide a headroom of 6 dB to accommodate the programme peaks. Note that this will often mean that the amplifier power rating is greater than the speaker power rating (unless your speakers are rated considerably higher than they need to be).

As a final check, be sure that the peak power rating of your speakers is high enough to cope with the peaks of the programme, and be sure that you do not abuse the headroom of the amplifier by driving the speakers at too high an average level. For more information see Power Ratings on the Amplifiers and Speakers page.

If the power requirement is high, the total power needed to achieve the needed sound level can be split between several speakers, provided that they are positioned and angled so as to cover the same audience area. If, however, multiple speakers are positioned and/or angled so that they each cover just a part of the audience area, then the required power input to the speaker (or group of speakers) covering each area will need to be considered separately if the audience areas are of different size or are different distances from their respective speakers. In large systems incorporating multiple speakers, several amplifiers are often used − or powered speakers are utilised.

For more detail on the selection of amplifiers and speakers see Amplifer and speaker selection for required sound level on the Amplifers and Speakers page.

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Q 6.
How should I set up my channel Gain controls?

A 6.
As each model of mixer is different, the only safe answer is "Set them according to the advice given in the mixer's operating instructions". If you don't have the instructions, they can often be downloaded from the manufacturer's website − see the Manufacturers page for links.

If for some reason you need alternative/additional guidance, here goes:

• If your mixer has individual channel metering, adjust the Gain for normal peaks to indicate around 0 dB to +4 dB (only green LEDs). Exceptional peaks should not exceed an indication of +10 dB (only green/orange LEDs). Red LEDs or LEDs marked 'Clip' or 'Peak' should never light.
• Otherwise, if your mixer routes the PFL signal to a level meter in the 'Master' section, engage the PFL switches one at a time and adjust the Gain control of that channel as above. If you think you need to make later adjustments to a Gain control always engage (just) the relevant PFL switch first, and watch the meter as you make the adjustment.
• Otherwise, if your mixer has both 'Zero level' and 'Clip' (or 'Peak') LEDs on each channel, adjust the Gain controls such that the 'Zero level' LED is lit when normal level is present but the 'Clip' LED never lights.
• Otherwise, if your mixer has only 'Zero level' LEDs on each channel, start with the Gain controls at minimum setting and then slowly turn them up until the LED lights only during normal peaks.
• Otherwise, if your mixer has only 'Clip' (or 'Peak') LEDs on each channel, start with the Gain controls at minimum setting and then slowly turn them up until the LED just lights only during exceptional peaks, then back off the controls just a little, so that the LEDs never light.

Remember to keep an eye on the channel levels (meters and/or LEDs) during the event. In particular, you may need to re-adjust Gain controls in the following circumstances:

• if the signal source level changes (e.g. if vocalists sing louder or closer to the mics, or if musicians play louder or adjust the settings of their instruments or associated on-stage equipment), or
• after you have made substantial changes to the channel EQ settings.

Also see Overload or level indication on the Mixing Facilities page.

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Q 7.
How should I set up my channel EQ controls?

A 7.
From an artistic viewpoint, the simple answer is "To get the best out of the sound from the source connected to each channel", or "To get as close as possible to the kind of sound that you want on each channel", or "To get the best fit of the sound of each channel in the overall mix". For an engineering perspective, see the Equalisation section of the Mixing Facilities page. For further advice on setting EQ, see the Mixing section on the PA Proverbs page.

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Q 8.
Should I use mono or stereo?

A 8.
This depends entirely upon whether the performers using the PA system, or the recordings that are to be played, rely on stereo sound to achieve their desired effect. If not, it is better to use mono because it is simpler and cheaper.

When using stereo, remember that a true stereo effect is only heard by listeners who are situated an equal distance from the left and the right speakers. The more unequal the distance, the less impressive the effect. This means that stereo works best in long, narrow rooms, and less well in short, wide rooms. Listeners whose location results in them effectively hearing only the sound from the speaker nearest to them (Haas effect) will probably get a less satisfactory sound than if mono were being used.

Back to the FAQ Contents.

Q 9.
How should I mike up a ...?

A 9.
There are two aspects to this:

• selection of a suitable microphone, and
• suitable placement of the microphone in relation to the sound source (microphone technique).

The best rule-of-thumb is 'experiment and find out what sounds best' with your available microphones, instruments and circumstances. And remember that there is no definitive 'right' and 'wrong' way of doing things in the artistic world. What sounds good to you, or is an ideal microphone technique to use in one situation, may be quite different to someone else's opinion, or to what would be best in different circumstances.

However, if you need a suggested starting point, or inspiration, or you just don't have the time to experiment, good sources of reference are provided by the top microphone manufacturers e.g. Shure.

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Q 10.
How do I wire up a microphone cable?

A 10.
If, as is usually the case, the cable is to have XLR connectors on both ends, then proceed as follows:

• Be sure to use a male XLR at one end of the cable and a female one at the other.
• First of all, thread the appropriate 'back end' connector parts onto both ends of the cable, making sure that they are both facing in the correct direction. In the case of some types of connector, this part will be the entire connector shell. But with other types it will just be the cable clamp and boot. As these parts may be different for the male and female connectors, take take not to now get the two ends of the cable mixed up.
• Prepare the ends of the cable by carefully cutting back the sheath for about 2.5 cm (1 inch) and then separating the screen (or drain wire, as appropriate) from the inner cores. Bare screen wires should be twisted together and trimmed back so that they are all the same length. Strip back the insulation of the cores by around 4 to 5 mm (just under a quarter of an inch), ensuring that all wires to be connected are the correct length to reach their respective terminals (see below). On some female connectors, the wire to pin 1 may need to be slightly longer than the others.
• Now solder the conductors to the appropriate terminals of the connectors (matching the parts previously threaded onto the cable), as detailed below. The pin numbers are usually embossed into the connector's insulation, sometimes on only one side. Take care to avoid dry joints and short-circuits between the terminals.
  • Connect one of the insulated cores (e.g. red) to pin 2 of both connectors.
  • Connect the other insulated core (e.g. blue) to pin 3 of both connectors.
  • Connect the cable screen (or drain wire, as appropriate) to pin 1 of both connectors. First placing some insulating sleeving over bare screen wires will help to avoid short-circuits.
  • Do not make any connection to the terminal that provides electrical contact with the shell. (For good reasons, it is generally accepted that the screen should be connected only to pin 1 − not to this shell terminal − meaning that the cable provides no signal earth connection to the shell. For example, this avoids possible earth loop problems if the shells of mated cable connectors touch earthed metalwork or other cable shells, and avoids nullifying the effect of earth lift switches on equipment into which the cable connectors are plugged.)
• Assemble the parts of the connectors, ensuring that the cable cores are not subject to stress and that the connector clamps engage with the sheath of the cable.

Q 11.
My amplifier has a quoted output power into 4 ohms, but my speakers are 8 ohms. Is this a problem and how much output power will I get?

A 11.
The figure quoted in ohms for your speaker is the impedance value of the speaker. This is a measure of how much it opposes the flow of current supplied by the amplifier. So, connecting a speaker with a higher impedance figure than is quoted for the amp will do no harm to the speaker nor to transistor-based amplifiers; a smaller amount of current will be drawn from the amp than the amp is capable of supplying.

However, the connected overall speaker impedance must never be less than the minimum value specified for the amplifier, as this would cause too much current to be drawn from the amp. (We say 'overall impedance' to cover the case where several speakers are connected to a single output of the amplifier. In this case you must work out the combined impedance of all the speakers that you are connecting, taking into account the way in which they are interconnected − usually in parallel.)

In contrast, amplifiers having valve-based outputs usually required a specific load impedance to be connected (no less and no more), to ensure proper operation and avoid damage. Some types have a switch that enables the required load impedance to be selected from a range of values.

Now we will consider how much power we will get. Since power is voltage times current, we have to consider both of these quantities. The voltage provided at the output of the amplifier, at a given point in time, is essentially unaffected by the load impedance (though the maximum output voltage attainable will be affected to some degree). However, as we have already said, the current is very much affected by the load impedance. From Ohm's Law, if the load impedance is doubled then the current will be halved.

So, with an 8 ohm speaker, half as much current will flow as with a 4 ohm one, causing half as much power to be obtained from the amplifier (given that its output voltage remains constant).

We can perhaps see this more clearly from an alternative formula for power: P = V² / R i.e. the amount of power is the voltage squared, divided by the impedance (or, more strictly, the resistive part of the impedance). When R is 8 instead of 4, we have to divide V² by twice as much, so we only get half the result.

Now in practical terms we are often concerned with the maximum power that we can get from the amplifier. In this case, we have to consider the maximum amount of voltage that it can supply, as well as what load impedance we are connecting. A higher load impedance will put less of a burden on the internal power supply of the amplifier, so in this case the amplifier will be able to supply a little more output voltage − and hence more power than you might expect from the simple theory above. For example, if an amplifier is rated at 200 W into 4 ohms, the theory would indicate that the maximum output power than can be obtained into 8 ohms would be half as much, i.e. 100 W. But, in practice, the reduced burden on the power supply means that the maximum output power into 8 ohms might be around 120 W.

So why is it so common for amplifiers to be rated into a 4 ohm load, when most speaker cabs are 8 ohms? This is because many users will want to connect more than one speaker to each channel of the amplifier. When plugging several speakers direct into the same channel of the amplifier, or plugging one speaker into the amplifier and then 'looping on' from another connector on that same speaker to a further speaker, then in practice you are connecting those speakers in parallel. In such cases, the maximum number of 8 ohm speakers that you can supply from each channel of a 'minimum 4 ohm rated' amplifier is two − because the overall impedance of two 8 ohm speakers connected this way will be 4 ohms. In this example, you would then get the full '4 ohm rated' power output from the amplifier − half of that value into each speaker. Of course this needs twice as many speakers, but that can be helpful because, for FOH, it gives more flexibility in how they are pointed, in order to get best coverage of your audience area. (Or in the case of monitors, it is often useful to have more than one speaker per monitor mix, whether for different performers or perhaps two speakers for a lead vocalist, for example.)

For more information see Impedance on the Amplifiers and Speakers page.

Back to the FAQ Contents.

Q 12.
Why do my speakers have different quoted maximum power values for RMS power and music power?

A 12.
'RMS power' is (in practice) short-hand for 'continuous average sine-wave power', and the rating is usually given on the basis of some standard test signal such as band-limited pink noise. The important word here is continuous − the test sound just goes on and on at the same level, so it puts a lot of stress (especially heating) on the speaker. That's why the RMS rating figure has to be lower than other kinds of rating.

But real programme material (whether music or speech) isn't like that. It has variations in level: peaks and quieter passages. So, when considering the rating of the speaker for music programme signals, the speaker can be rated at a higher value than for the continuous test signal, because it only has to stand that maximum amount of power for short intervals − on the peaks of the music. For more information see Power Ratings on the Amplifiers and Speakers page.

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Q 13.
I changed my speakers − why do they sound louder (or quieter) than my old ones, even though they have the same impedance value?

A 13.
Since they have the same impedance as your old ones, they will be drawing the same amount of audio power from your amplifier (see Q 6) − assuming that you haven't changed any settings of your equipment. But, a given power level into the speakers does not directly indicate the sound level they will produce − that depends on how efficient your speakers are at converting electrical power into sound. We call this 'efficiency' value the sensitivity of the speaker, and it can vary a great deal between different makes and models of speaker. So, 100 W of power into one type of speaker can easily sound like 200 W or even 400 W of power into another type! For more information see Speaker Sensitivity on the Amplifiers and Speakers page.

Back to the FAQ Contents.

Q 14.
How do I connect the balanced output from my mixer to the unbalanced input of my power amp, without getting hum problems?

A 14.
This is a common situation when using semi-professional amplifiers, and may well need extra care to avoid problems with hum. Firstly we have to say that the ideal solution would be to use an amplifier having balanced inputs (and, preferably, a signal-earth lift switch). But if that isn't practicable...

When using good quality screened cable (such as a standard multicore), an unbalanced interconnection between the mixer and amplifier can often be satisfactory − depending on the distance involved and the amount of magnetic and radio-frequency interference present in the vicinity.

However, there are two main ways that hum can be introduced into an unbalanced interconnection, even when it is well-screened:

• When both the mixer and the amplifier have a safety-earth connection, a small mains-frequency current may flow in the screen of the mixer-to-amplifier interconnection cable (an 'earth loop' currrent), which can result in an audible hum.
• Hum induced into the interconnecting audio cable from nearby mains-powered equipment and mains cables.

Now provided that your amp is of the 'double insulated' (also called Class II) type, designed to be safely operated without a safety earth connection (i.e. is fitted with a 2-core mains cable), you should not be troubled by the first of these hum sources. (However, note that other equipment connected at the amplifier end of the interconnection − such as an active crossover or a graphic equaliser − may have a safety earth, or the chassis of the amp or other equipment may be in contact with earthed racking.) And, provided you have laid out your cables and equipment so as to keep mains and audio as separate as possible, the second hum source is often also not a problem. This is because the signal level out of the mixer is relatively high (much greater than the signal level from microphones or guitar pick-ups, for example), so any hum induced into the mixer-to-amplifier interconnection is much less significant than if the same amount of hum were induced into a microphone-to-mixer interconnection, say. (This is why mic-level interconnections are pretty much always balanced.)

Hence the statement above, that unbalanced interconnections can often be satisfactory. To make such an unbalanced interconnection, the cable screen should interconnect pin 1 of the multicore XLR with the sleeve terminal of the jack input to the amplifier, and the inner screened core of the cable should interconnect pin 2 of the XLR with the tip terminal of the jack − leave pin 3 of the XLR (or any core connected to it) unconnected. (This will work in 99% of cases, but some small older mixers of US origin have a semi-balanced output with pin 3 'hot' instead of pin 2 − when using these you will need to connect XLR pin 3 to the tip terminal and leave pin 2 unconnected.)

However, if your amplifier is of the Class I type, which require a safety earth (3-core mains cable), then there will be an earth loop through the audio interconnection from the mixer, and this loop might cause some noticable hum − especially if the amplifier is fed from a different mains outlet from the one used for the mixer.

N.B. Never disconnect safety earths to resolve earth loop problems, as this will create a potentially lethal electric shock hazard.

So to be sure of avoiding hum problems in the Class I amplifier situation, you need to effectively adapt the input of the amplifier to accept a balanced feed, and have the facility to break the earth loop by interrupting the signal earth path (NOT the mains earth path!). When doing this, it is best to have a balanced cable run to the amplifier location, and to make the adaptation to unbalanced at that end.

The best way of doing this is to use the tool made for the job − a line-level balanced-to-unbalanced convertor unit (sometimes called an 'earth isolator' or a 'ground isolator'), with a 1:1 impedance ratio. These are available ready-made, but professional quality units are fairly expensive. (I wouldn't recommend the very cheap ones.) You can also make them yourself quite easily, if you're into that sort of thing, using a suitable high quality audio transformer that is designed to carry the required level. Make sure that the unit has an earth-lift switch, which you should set to the 'Lift' position to break the connection between the signal earth of the audio interconnection and the safety earth of the amplifier. (Note, however, that if you are using an earth isolator unit with a Class II amplifier then the earth-lift switch may need to be set to the 'Earth', or 'Ground', position.)

Alternatively, it can be done using a DI box 'in reverse' (i.e. balanced in, unbalanced out). But it must be a passive DI box − i.e. one which uses no power source (battery, mains, or phantom power). The passive types achieve the balanced/unbalanced conversion by using an internal audio transformer (rather than electronically), and such a transformer operates happily 'in reverse'. The DI box needs to have an 'earth lift' facility (as most of them do), which, as for the earth isolator unit, should be set to the 'Lift' position. Set its pad switch (if it has one) to the '0 dB' position.

However, there are two possible problems with the DI-box method. The first problem is that DI boxes are generally designed with low-level signals in mind, so (to keep the cost down) they often use transformers which are not intended to carry the high level that you get from a mixer output. The result of putting this high level through such a transformer would be some distortion of the signal − possibly just on the signal peaks. (This will not harm the transformer, though.) So be sure to use a good quality DI box, as these are more likely to have good quality transformers. Also, try setting the amplifier input level control at or near maximum − this means that, for a given output from the amp, it needs less drive signal from the mixer than if you had set its input level control to a lower setting, hence less chance of exceeding the level at which the transformer would start to introduce distortion. (Be sure, though, that the power rating of your speakers is appropriate for the amp's maximum audio output power, just in case you accidentally drive the amp with a full-level signal.)

The second problem is that as DI boxes are intended as impedance matching devices, they intentionally do not have a 1:1 impedance ratio. So, when used in reverse, their output impedance will be relatively high and there will be an increase in signal level. However, unbalanced inputs of amplifiers are likely to be fairly high impedance (compared with balanced inputs), and so may not unduly load the output of the DI box. And the increase in level can be compensated for by simply reducing the output level from the mixer. (Don't use the pad switch of the DI box for this purpose, as the pad is not designed to be used 'in reverse' and would probably further increase the output impedance of the DI box.)

In the few instances where the impedance change and/or level change from a DI box is a problem, you could try using two identical passive DI-boxes 'back-to-back', i.e. link their jack connectors with a short jack-to-jack cable. Connect the multicore feed from the mixer to the XLR of one of the boxes, and connect the XLR of the other box to the amplifier input jack as follows: pin 1 and pin 3 of the XLR to the sleeve of the jack, and pin 2 of the XLR to the tip of the jack. Set the earth lift switch to 'Lift' on only one of the boxes − it shouldn't matter which. Ensure that both of the pad switches are set to '0 dB'. This arrangement should eliminate the problems of using a single DI box, because the two boxes will cancel each other's impedance and level changes.

Finally, for more general information on the causes and avoidance of hum, see the FAQ entry How can I solve a hum problem?.

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Q 15.
How do I calculate ...?

A 15.
Here are some common formulae for audio-related calculations. The symbols and units used are:

• I  for current in amps
• V  for voltage in volts
• R  for resistance in ohms
• P  for power in watts
• VA  for apparent power in volt-amps
• dB  for decibels
• f  for frequency in Hz
• Z  for impedance in ohms
• v  for velocity in metres per second
• T  for temperature in degrees Celsius
• λ  for wavelength in metres
• The subscript '_ref' indicates a reference value.

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This page last updated 17-Jan-2012.