Sound Reinforcement (PA) - A Guide to Making Live Sound Work For You

Discussion in 'Bad Dog Cafe' started by simoncroft, Aug 2, 2019.

  1. simoncroft

    simoncroft Tele-Meister

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    Show me assigned!

    Although ‘routing and grouping’ sounds like something from a Fifties song (“C’mon everybody, we’re a-routing and a-grouping…”), it’s an important area if you are going to get a handle on how signals are moved and managed on a mixing console.

    With that aim, we’re going to use Diagram N as our reference. It shows the mic/line input (1), subgroup, or group (2) and main mix output (3) strips from a Soundcraft GB2 console. Other mixing systems may not be identical, but the fundamental ideas can be transferred to almost any desk. It’s the areas marked in black we’re concerned with today.



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  2. simoncroft

    simoncroft Tele-Meister

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    Alongside the fader on the input channel there are a number of push buttons. The first of these is marked Mute and has a LED next to it to show when it is in use. There are a number of instances where the Mute function can be useful, but for now let’s just say it “does what it says on the tin”.

    Our main focus today is the group of three buttons mare 1-2, 3-4 and Mix. As you’ve probably worked out, these buttons route the input channel to three possible destinations. These are (sub) Group 1&2, Group 3&4 and the main stereo output, also referred to as Mix. In all instances, the Pan control determines how much of the signal from that channel goes to the Left/Odd-numbered outputs and how much to the Right/even-numbered outputs. The main significance of this is that pairs of subgroups can contain a complete mix stereo. We’ll look at this again when we come to the Width control on the subgroups.

    Also on the input channel are LEDS for Peak level (PK) and Signal Present (SIG). These are useful aids when initially setting the input levels. The PK LED will also warn of overloading and help to identify the source of feedback.

    Group channels on this particular console have a Width control. The Width control is basically two pan controls in one, so when it’s turned all the way clockwise, the effect is the same as having one group panned hard left and the other hard right, which is why the position is marked Stereo. With the Width turned all the way counter-clockwise, it’s the same as two pan controls pointing dead center, which is why it’s marked Mono in this position.

    This is by no means a standard inclusion but it is quite useful, as this example will hopefully show you. Let’s say you’ve decided to send all the drum mics to Groups 1-2 and you’ve already panned the individual channels to reflect the position of each drum on stage. There’s a good chance that by the time this mix comes out of the main rig, it’s way too wide. By that I mean, when the drummer goes round the toms, the spread is about 20ft, whereas if you were standing in front of the kit, the actual spread would be about 3ft. Reducing the Width control until the stereo field is more realistic is a lot easier than adjusting all the individual channels (which is basically the whole point of subgroups!).

    The three buttons on the Group channel – Mix and two PFLs – are relatively straightforward but you might want to refer to the box I’ve called ‘What’s PFL again?’ As the name implies, the Mix button routes a pair of Group channels to the main mix.

    You might wonder why the groups are not simply hard-wired to the Mix output. After all, if you can’t hear them in the main mix, what use are they? There are a few reasons why you might not want your subgroups to go to the main mix. One of these is if you are also making a multitrack recording and want to use the group faders to set the optimum record levels, without having to consider what that does for the main Front of House mix.

    The main Mix fader (on some desks you’ll find two for Left and Right) has no associated routing or PFL because the signal goes straight to the main rig.
     
  3. simoncroft

    simoncroft Tele-Meister

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    ** What’s PFL again? ************
    We’ve touched on the function of PFL before, but let’s look at it in slightly more depth. The Pre Fade Listen button allows the engineer to hear a channel in isolation (as well as checking its level on the main meters). This can be useful when, for instance, trying to EQ just the snare drum. You could ask the drummer to keep playing only the snare but using the PFL makes it easy to hear the snare, while keeping it in the context of how the drummer will actually play the whole kit during the set.

    Up-market mixing consoles have a related function called Solo. This is superior to PFL in that Solo allows the engineer to hear a channel with the pan setting and any effects in place (hence the control is sometimes referred to as ‘Solo in place’). While Solo is generally more useful than PFL, it is also more complicated and expensive to engineer on an analogue console. On digital desks, the engineering issues are less relevant from a design perspective, because there is no difference in component costs between PFL and Solo.

    AFL stands for After Fade Listen and is functionally very similar to PFL but is generally found on the Master section of the mixer, where it permits monitoring of Aux mixes and other functions.

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  4. simoncroft

    simoncroft Tele-Meister

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    In this next post, I’m including a short section on The Matrix, which is nothing like as exciting as the name might imply. In fact you can live a long and fulfilling life without ever using it, but I don’t want anyone to think I’ve glossed over a few controls because I don’t know what they do either!

    We should also take a good look at all the sockets provided on a mixing console. There are a lot of them and understanding what they are for is a big part of getting the most from a console.

    After that, let’s consider ourselves done with mixing consoles for a while and turn our attention to the Sound Reinforcement gear we find on stage: microphones, DI boxes and the stage box.

    The Matrix!

    Actually, this post isn’t just about the Matrix, it’s about all sorts of disparate stuff you find on the Master section of bigger mixing consoles. It’s all useful stuff but looks kinda intimidating if you’re used to working with smaller desks that don’t have these features.

    Diagram O is what we’re working with today. The first two channel strips are part of the Group outputs labeled 2 in Diagram N in my previous post. The first thing to understand is that the remainder of these channels strips shown in Diagram O has nothing to do with the function of the group faders below them.

    Sometimes, when designing a mixing console, you have to put things where there is space. That’s why I described this section of the mixer as ‘disparate’. Perhaps ‘administration and communications’ might be a better way to describe what goes on in this particular part of the desk.

    The first seven rotary controls to both group fader strips are almost identical, and they are essentially two separate 6-into1 mixers. What makes these little mixers special is that their inputs are derived from the four (sub)group outputs and the main mix outputs. If you look at the markings next to the controls, you’ll see that this is the case. It’s only when you get to the seventh rotary that they are different: the one on the left is marked MTX 1, while the one on the right is marked MTX 2.

    This is because the first Matrix mix goes the MTX 1 output on the back of the console, while the second goes to MTX 2. But what you use these mixes for? Well, one simple scenario is that you use them instead of, or in addition to, the monitor mixes you could create using the Aux mixes. Assuming Groups 1-2 contain the drum mix, while Groups 3-4 contain the vocal mix, there is the scope for two monitor mixes that have more drums and/or more vocals than the Front of House mix. That’s going to keep a lot of bands happy, and makes life very simple for the sound engineer.


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  5. simoncroft

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    There are other uses for Matrix mixes, depending on the kind of production, and the demands of the venue. For now, it’s probably enough that we grasp what the controls on the console do. Note that the Matrix mixes are provided with AFL and Mute buttons.

    Below these, we have the master levels for the various Aux mixes, again with AFL monitor buttons. In the case of pre fade mixes, these levels will be the master volumes for on-stage monitor mixes, while the post fade mix levels will control the signal level going to outboard effects such as reverbs.

    Moving to the Mix channel strip to the right of Diagram O, you’ll see that there are two sockets. One of these is for headphones, the output to which is mainly determined by whether or not any PFL/AFL buttons are engaged.

    The other socket is a 3-pin XLR for a Talkback microphone. In most performance situations, the talkback mic will be used as a way for the Front of House engineer to communicate with the stage. That is the reason for the Aux 1-4 button just below the Talkback volume control.

    The ability to send the Talkback mic to the Group outputs is probably more useful in theaters, but there is one (possibly unique) use I swear is true. There used to be a very good band in the UK called The Sensible Jerseys. Their Front of House engineer was also a harmony vocalist on some of their numbers. Presumably, he routed Talkback to Groups in order to be part of the mix. Whatever, it was very strange to see the guy behind the desk joining in.

    Also in the Master section, you’ll find the main meters. Typically, you’ll also get a way to add a stereo player (CD, MP3 etc) for music between sets.

    (This is just a short post until we do the whole “what are all these sockets for” thing next time.)
     
  6. simoncroft

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    The Function of the Inputs/Outputs on a Mixing Console

    Most people probably think the same thing I did when I looked round the back of a full-scale mixing console for the first time: “Oh no! This looks really complicated!” What I didn’t realize until I got to grips with all these connections was how many creative possibilities they could open up. Live multitrack recording on a desk that doesn’t look up to the job is just one of the possibilities.


    Fortunately, just like the channels strips on a console, the connectors on the back are mostly duplication. Take a look at Diagram P and you’ll see that the whole console can be reduced to very few component parts. The best way to explain what the sockets do (or more correctly, the things you can use them for) is to start at the inputs on the right and work our way through until we get to the outputs.


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    ****** What is a TRS jack?*********

    TRS stands for Tip, Ring, Sleeve and a jack plug of this type has a number of applications. The most common is on stereo headphones, where the Tip carries the Left channel, the Ring carries the Right and the Sleeve is the Common. On pro audio gear, TRS jacks are frequently used for a couple of reasons. One is to allow line level connections to be balanced (that is to say, the audio phase and antiphase connections are kept separate from the Ground or Screen, as shown in Diagram G in an earlier post). The other common use is to create send-and-return Inserts that function as both input and output. Both of these functions could be performed using XLRs, but XLR connectors are more expensive and take up more room than jacks. In addition, using a different connector for the Mic and Line level inputs makes it impossible to mix them up.

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  7. simoncroft

    simoncroft Tele-Meister

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    Mono Mic/Line input channels

    These take up the majority of the real estate, so anything you learn reading while this section can be used over and over. The socket you’ll be using most often is the 3-pin XLR Mic input. These provide for noise-cancelling ‘balanced’ connection, and also allow 48V Phantom Power to be fed to condenser mics. (Phantom power can also be used on active DI boxes. There is no problem with using the Mic input for this purpose, but the Gain control will typically need to be set lower.)

    Next to this is the Line input. Although this goes through the same preamp section on the desk as the Mic input, it is more typically used for sources such as electronic keyboards, submixers and outboard processors.

    Another common inclusion is an Insert. This sends a signal from the preamp stage of the mixer on the Tip of a TRS jack and returns it on the Ring connection. A typical use for this would be to include a compressor, or other dynamic device that is only required on that channel.

    NB – Occasionally a TRS insert on a mixing console is used in conjunction with an outboard device that is also fitted with a TRS jack for send/return. It is worth bearing in mind that the cable must be reverse wired, so that the send (output) of the mixer goes to the return (input) of the device, while the send (output) of the device goes to the return (input) of the console.
     
  8. simoncroft

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    The Direct Output is again balanced on a TRS jack and has an associated Pre switch. Together they are a very useful inclusion if you need to feed either a multitrack recorder, or a separate monitor mixing console. By default, the Direct Output is post fader, meaning its level is controlled by the channel fader. You would probably consider working this way if your intention was to lay tracks to a recorder and there was no Front of House mix required. Conversely, if your main need is to create a Front of House mix, while feeding a multitrack recorder or monitor mixer at the same time, pressing the Pre button switches the output to pre insert point. This means that once the input gain is set, the Direct output will not be altered by any changes to the main mix.

    One of the cool things about taking direct input channels to a multitrack is that you can feed a lot more separate recording tracks than you have subgroups. To put it another way, there are plenty of desks out there that have, say, 16 input channels. But if you want to buy a desk with 16 group outputs, you are in a very different price category. So direct out give you a lot of recording power, with having to use a big-budget desk. But what if your otherwise perfectly functional 16 channel desk doesn’t actually have direct outs? That’s where you can benefit from some creative thinking about how connections can be made.

    A lot of desks that don’t have direct outs are equipped with insert points. If you wire a TRS jack so that the Tip and Ring are shorted together, you can take that as your signal and the Sleeve remains as shield. It isn’t balanced, and the quality may not be quite as good as a dedicated Direct Out but it works.
     
  9. simoncroft

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    Stereo Mic/Line input channels

    No prizes for guessing the main difference between these and the Mono channels! Yep, on these channels there are Left/Right Mic inputs and another two for Line inputs. The Stereo channels are useful for several different types of input source, including sub-mixes from DJ decks or keyboard rigs, stereo mics on drum overheads and other percussion, and external effects returns. In short, any kind of source where it makes more sense to deal with the incoming signal as two halves of a stereo pair, rather than separate mono signals that are likely to need different EQ settings and levels.

    This particular mixing console has a very neat switch called Line To Mix/Line To Chan(nel). In the up position, the Mic inputs go through the entire channel strip but the Line inputs go through only a separate Line level, then straight to the main mix. That means you can use the main channel strip for a stereo pair of mics and still use the Line inputs for effects returns.

    In the down position, the Line To Mix/Line To Chan(nel) routes the Line inputs through the channel strip and disables the Mic inputs for that channel completely. This is the more appropriate setting if you have an incoming submix from stage that is likely to need EQ, monitor and effects mixes.
     
  10. simoncroft

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    Aux outputs

    These are to the right of a section I’ve marked ‘Outputs and Auxes’ on Diagram P (above). There are six Aux mixes on this console, so six Aux outputs. So why are four of them on XLRs and two of them on TRS jacks? Well, the default configuration of the GB2 is four pre fade auxes and two post fade, which implies four going to monitor amps and two going to external effects units. (The two middle aux mixes can be switched to post fade but you’d hardly expect the output connectors to switch from XLR to jack…)

    Group and Matrix outputs

    There are four of these on XLRs, and each one has an accompanying Insert on a TRS jack. They would typically be used for putting one compressor on the drums and another on the vocal mix, for instance.

    The outputs for Matrix 1 and Matrix 2 are below the Auxes. If you have read earlier posts in this thread, you’ll know that Matrix mixes are made up from the Group and Mix outputs, which makes light work of providing two alternative monitor mixes, as each one has only six levels to mix. In small venues, the Matrix mixes would be quite acceptable as the only stage monitor mixes. In large venues, it is more likely that the Matrix mixes would be used for general side-stage ‘fills’, while the Aux mixes were used to provide individual artists with specific mixes. Which way you go with this is partly down to have much stage area there is, now many monitor amps and speakers you have available, and how much time there is for setting up monitor mixes.

    Main Outputs

    As the name suggests, Mix L and Mix R are the main stereo outputs. Again these are equipped with insert points that could be used for a compressor. (You may be wondering if it wouldn’t be easier to simply place the compressor after the main mix outputs, but the insert points are pre fader. Therefore, moving the output fader(s) does not affect the compression threshold. Simply routing the outputs to a compressor would mean there was more compression applied when the faders were up high and less as the output level was reduced.)

    The Mono output below this is simply the Left and Right Mix outputs summed. One possible use is to feed dressing rooms, so that artists can hear what’s going on, so make themselves ready in time.

    2-Track and Monitor

    The RCA jacks (‘phonos’ if you’re a Brit) are designed to connect a 2-track recorder, which can be used both for music between acts and to record the main mix. The signal to the Monitor outputs feed the control room monitors in a recording situation, or the sound engineer’s monitor speaker in a live set-up. This is normally the same signal as goes to the Headphone socket. Once we get into actually operating the mixing console, the way the monitor system is used with PFL/AFL and other routing options will be explained.
     
  11. simoncroft

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    Multiway I/O on a Mixing Console

    Before I launch into discussing the sort of connectors and protocols that can support multi-channel transmission on a mixing console, it probably helps if I set the scene a little. The Soundcraft console we worked through above uses XLRs and TRS jacks almost exclusively, so why would you need anything else? One reason is that, as you work larger venues, it makes more sense to put all the on-stage mics and DI boxes though a single cabling system, rather than using individual cables.

    Another reason, here in the digital age, is that the data we are transmitting around the system does not have to be limited to audio – it can also include control data. That has some really cool implications. For one thing, it means that all the amplifiers can be located where they belong, which is on the closest possible cable runs to the loudspeakers. Thanks to networking technology, we can now control the amp levels from almost anywhere, so we don’t need them situated close to the mix position.

    Equally, now we have Wi-Fi and tablets as almost standard issue, every aspect of system set-up can potentially be controlled from any location, whether that means system EQs, power levels to various speakers around the venue, monitor mixes… While this IT-based approach to system control is not new, it used to affordable only on the highest level of concert tour and installed venue systems.

    Because the handheld devices most of us can afford are now so powerful, it is cheaper for an audio manufacturer to provide high-level DSP (Digital Signal Processing) functionality in a rack device that interfaces with iPads than it is to make a large-scale analogue mixing console.

    Does that mean the hardware-based mixing console is “dead”? No! Trying to manage the sound for a live event using only software control makes about as much sense as trying to drive a car using an app on your phone. In both situations, we need a predictable physical ‘interface’ that our eyes can understand at a glance, but our body can control without even looking.

    What it DOES mean is more-and-more ‘smart’ functions entering Sound Reinforcement systems that are affordable to ‘the rest of us’. Before we can get into that, we need to get back on-topic and look at some of the I/O (Input/Output) connectors you may find on a mixing console.
     
  12. simoncroft

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    Analog Multicore

    There comes a point where the size of venue makes it inevitable that a ‘multicore’ cable and stagebox makes more sense than individual mic cables. That is also the point at which it makes sense to question whether it might be more efficient to use a multiway connector that could carry all channels, rather than a multicore cable that terminates is a series of ‘tails’ with individual XLRs at the console end of the business. If you’re not familiar with the type of multicores I’m talking about, here’s a page full of ‘em https://www.proaudiocentre.com/shop/audio-equipment/stage-boxes---multicores/index.html

    One of the more common multiway connectors is known as an EDAC. With their rectangular shape and central locking bolt, they are easy to identify. Some examples are here. http://www.canford.co.uk/EDAC-MULTIPIN-CONNECTORS

    However, there are significant disadvantages to using analog are a transmission technology over long distances. These include the cost and bulk of the cable, as well as the relative inflexibility of analog compared to digital when it comes to transmitting control data along with the audio.

    Digital

    Almost any standard developed by the computer industry for data transmission has potential use for professional audio. Two of the most obvious examples are USB and FireWire (also known by the snappy name ‘IEEE 1394), both of which are popular choices in home recording systems and smaller SR applications. More on USB here: http://en.wikipedia.org/wiki/USB More on Firewire here:

    Cat5, and its variants such as Cat5e and Cat6, which are typically used to carry data over LANs using Ethernet can also be used in pro audio. More here: http://en.wikipedia.org/wiki/Category_5_cable http://en.wikipedia.org/wiki/IEEE_1394

    It’s flagging up at this point the distinction between connector types and the data standards, or protocols, of the information they carry. So although a USB connector will always be used to carry one of the USB standards, and the same can be said of FireWire, a Cat5 cable may be carrying any of a number of networking standards, including Audinate’s Dante system (on which more below). Similarly, a 3-pin XLR can be used to carry AES3 (aka AES/EBU) digital audio, rather than analog. Although this makes for a confusing world, all we can do as users is make sure that the equipment we are attempting to interconnect is using compatible standards!

    For instance, the ADAT Optical Interface (‘Lightpipe’) uses the same fiber optic connectors as TOSLINK (domestic) and S/PDIF (semi-pro audio) but is not data compatible with them. Lightpipe was originally developed by ADAT as a way of connecting its digital multitrack tape recorders but was adopted by other manufacturers as a handy way to transfer eight channels of audio through a single connector. Because there is no electrical connection, a further advantage is there is no risk of Lightpipe causing a ground loop.

    Moving back to Dante, there are several really cool things about it. Technically, it’s excellent, having been developed as a multichannel audio networking technology specifically for audio applications. But it is based on IT standards, so equipment such as Ethernet routers can be used to configure the network. It can also be used with Cat5e, Cat6 and fiber optic.

    Possibly the best thing of all is that a lot of major pro audio manufacturers have decided to license it. So if you want to integrate equipment from Rane, Nexo, LabGruppen, Klark Teknik, QSC, Yamaha and Shure (to name just a few) you can do so. More here: www.audinate.com

    Going back to the core topic of audio connectors (and sorry, I’ve brought in a lot of new ideas here) USB, FireWire, Cat6 and S/PDIF connectors are very useful but they do not have the rugged reliability and mechanical locking of connectors used in Sound Reinforcement, such as XLRs and SpeakONs. http://www.neutrik.co.uk/en-uk/speakon/

    The connector family that looks to become the ‘fiber XLR’ is known as OpticalCON. It’s worth taking a look because there will be more and more of these round the back of the rack.

    http://www.neutrik.co.uk/en-uk/audio/opticalcon/opticalcon-advanced/

    http://www.markertek.com/category/opticalCON-Connectors
     
  13. simoncroft

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    Microphones – An Introduction To Specs and Real-Life Performance

    As guitarists, we know that putting a better set of pick-ups into a guitar can make a lot of difference to the sound. It’s a swap most of us make at one time or another, either because we want to upgrade an affordable guitar so that it sounds like a top-line instrument, or because we are chasing a particular ‘signature’ tone.

    Let’s turn this round the other way for a minute and imagine we own a really wonderful guitar, but it has no pickups. (Why would that be? I dunno. If it helps, let’s say I stole the pickups. It’s only an imaginary guitar, so when you set the imaginary cops on me, I’m sure I can talk my way out of it…) Whatever the reason, you now have a Custom Shop dream guitar without pickups. Would you: a) go out and buy the best pickups you could afford, b) the cheapest pickups you could find, or c) leave it without any pickups at all, and hope you can borrow some?

    As this is a guitar forum, I confidently predict that you will go straight for Option A, and quite rightly too. But often the same musician who went for Option A on their guitar will settle for Option B or C when it comes to microphones. Typically, that’s not the musician’s fault. It’s just the mics they end up with are maybe the ones supplied by the venue, or the best they could afford to buy.

    The message I’m trying to put across in this post is: microphones are pickups for voices and acoustic instruments. Choosing microphones of good quality is an important starting point, especially if you can afford it. But knowing the right ones to choose for the job – and how to position them – is the key to getting the best possible result, whatever the budget.

    Before we get into specific makes and models – or even the general types of mic you can encounter – let’s take a look at the characteristics we might want from a ‘perfect’ stage microphone. This will help us to get to put the ‘paper specifications’ for mics into some practical context.
     
  14. simoncroft

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    Screen Shot 2019-08-10 at 15.57.04.png

    Frequency response

    Diagram Q shows a series of frequency plots from 20Hz-20kHz. You can read plots of this type very much as you would the settings on a graphic equalizer: the low frequencies are on the left of the graph and the high frequencies are on the right, with the ‘volume levels’ shown in Decibels from lowest at the bottom to highest at the top.

    The frequency range of 20Hz-20kHz is used for most microphone specifications because it is the same range a CD is capable of reproducing and is nominally the range of human hearing. The reality is slightly different, in that very few adults can hear anywhere close to 20kHz, especially if they have beer inside them. Also, most Sound Reinforcement systems have little frequency response below 40Hz. And even those rated up to 20kHz will typically be -10dB* (about ‘half volume’) that high up in the frequency range.

    *******************

    *I don’t want to get too sidetracked into the relationship between Decibels and loudness, but as a rough rule of thumb, a 10dB increase would be perceived as being twice as loud, and a 10dB decrease (-10dB) would sound half as loud. There is very good information about Decibels here: http://www.sengpielaudio.com/calculator-levelchange.htm

    And here: http://trace.wisc.edu/docs/2004-About-dB/

    *******************
     
  15. simoncroft

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    The frequency response limitations of the speakers are good news when considering SR/PA applications because high-quality condenser microphones that can come close to the theoretical ideal of 20Hz-20kHz and can stand up to use on stage don’t come cheap. The horizontal red line in Diagram Q is the best possible ‘flat’ frequency response across the full range. The yellow line is based on the actual frequency plot for a microphone that costs about $3,000 (£/€2,000). The solid blue line is more like the response you can expect from a stage vocal mic costing typically $100 or less.

    So does that mean that the only way to get top quality reproduction when using 10 mics on stage is to go out and blow $30,000? No! Here’s the reason why.

    The frequency response of the stage vocal mic (blue line) appears limited but is actually quite useful for the job it was designed to do. For one thing, there is almost no useful sound from the human voice below 100Hz, so the fact that our vocal mic slopes off towards that range helps it to avoid picking up unwanted sounds, such as stage rumble, plus unwanted spill from the bass guitar and the low end of the drum kit directly.

    Generally speaking, the most important frequency range for the human voice in terms of character and intelligibility is between 4kHz and 8kHz, which is where most stage vocal mics exhibit a ‘presence peak’. This helps vocals to cut through the mix and does not diminish perceived vocal quality unless taken to extremes.

    From the microphone designer’s point of view, lifting the frequency response in the ‘presence’ range also helps to ensure a usable amount of response at the frequencies beyond 15kHz. This is the frequency range where vocal ‘sibilance’ is experienced – ‘ess’ sounds and a breathiness that lends sparkle to vocals.
     
  16. simoncroft

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    But not everything we’ll want to mic up on stage is always a singer. Vocal mics are also useful on electric guitar speakers, snare drums and toms, plus brass-wind (just 'brass' in Europe) depending on the sound you are after. They are not so good in any application where extended low or high frequency response is an advantage.

    On a kick drum (bass drum, to us in the UK) a flat frequency response is not very important. So although you can get very good results using a studio-quality mic with a full frequency response, that’s an expensive way to go. Most mics designed for kick drum have a peak around 50Hz, to ensure a solid low-end ‘thud’ and another somewhere in the 5kHz-8kHz region, where the ‘slap’ of the beater hitting the head can be found. What happens in the frequencies between those two points is not so critical, which is why many kick drum mics have a frequency plot that looks like a mountain range.

    Bass guitar is a different issue. Although the fundamental notes on a 5-string bass lie between 30Hz and 98Hz, the overtones and harmonics form a very important part of the sound. As a result, it’s not safe to assume a microphone is ideal just because it has an extended low end. In a lot of situations, a DI (Direct Injection) box, or a direct output from the bass player’s amp, will give better results. Double basses benefit greatly from a microphone with a wide frequency response, but a transducer or pickup can be the more practical option if the player has already fitted one to the instrument. The best solution greatly depends on the situation and the sound the act is aiming for.

    A similar pay-off exists with acoustic guitars, and indeed all other acoustic stringed instruments. It is extremely difficult to balance traditional orchestral instruments in the context of a rock band, due to the differential in volume levels. In those situations, transducers and pickups can be the only way to go. But for pure sound quality, microphones will always be the way to go – and the closer they are to the ideal ‘flat line’ response, generally the better the results will be. (In real-life situations, it can still be a good idea to shed some low-end response but that idea can wait until another day.)

    Going back to percussion: cymbals, including hi-hats, are literally at the other end of the spectrum to bass instruments. Here, extended high frequency response is everything, so a pair of studio-quality mics forming stereo ‘overheads’ for most of the cymbals, and another specifically on the hi-hat, makes for a good starting point.

    Similarly, when miking an upright or grand (ie ‘acoustic’) piano, the sheer range of the instrument means that only mics with the flattest, most extended, frequency responses will do full justice to the instrument.

    As a brief summary: the aim is to produce a full frequency mix, full of any unintended coloration. Because different instruments operate in different parts of the frequency spectrum, we can achieve our aim without using a full-frequency mic in all vocal and instrument applications. In fact, microphones with limited frequency response can be helpful if they exclude parts of the frequency spectrum that are not meaningful for the voice or instrument we want to amplify.


    However, frequency response is only one criterion for selecting stage microphones. In my next post, I’ll look at other issues such as polar patterns, and sound pressure levels. (I’ll also explain what the dotted blue line on Diagram Q is all about…)
     
  17. simoncroft

    simoncroft Tele-Meister

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    Microphone Polar Patterns

    In live sound applications, we tend to use ‘directional’, or ‘unidirectional’ microphones almost exclusively. To put it as simply as possible, these are mics that pick up loudest from the front, less at the sides and hardly at all from the back. This is how it is possible for singers to stand in front of loud stage monitors without howling feedback (which is exactly what happens when mics pick up the sound from the speakers and then sends it to the amplifier, back through the speakers…)

    However, there are inherent limitations to directional microphones, and it is useful to understand what those limitations are. Not only will it help you to use mics more effectively, it will also help you to understand why mics that might look similar on paper can be quite different in real-world use.

    Diagram R contains a lot of information, so let’s go through it a step-at-a-time. First off, what is a ‘polar plot’? It’s a little map that shows you, by points of the compass, how loudly a mic picks up from the front, the sides, the back – and all points in between. For very expensive microphones, the frequency and polar plots supplied are computer-generated readouts of the actual unit you are buying. But for most mics, the plots are generalized descriptions of what the manufacturer is shooting for. Part of what I hope to do in this post is help you to understand how much information you are being given, so you have a better idea of what you’re buying into.


    Screen Shot 2019-08-11 at 16.54.02.png
     
  18. simoncroft

    simoncroft Tele-Meister

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    Let’s start by looking only at the red lines, which represent measurements made at a frequency of 1kHz. The mic on the left has a perfectly circular plot at 1kHz, which tells us that it picks up at the same volume level all the way round. (Think of this as a perfect sphere, rather than a circle, because the 2D plot is actually giving us information about what’s happening in 3D space.) A microphone that has this pick-up characteristic is called ‘omnidirectional’.

    The mic on the right has a plot that shows far less pick-up to the sides and the back than at the front. This tells us that the microphone is to some extent directional. The most common type of directional mic is called ‘cardioid’, because the plot looks somewhat heart shaped. However, ‘cardioid’ is not one absolute standard. There are ‘wide’ cardioids, ‘tight’ cardioids, and as I’ll attempt to show you, cardioids that are plain all over the place!

    Very often, when you go to buy a mic, the manufacturer has placed a handy graphic on the packaging that looks very like the red-line plots we’ve just been examining. While this is useful for telling you that one mic is omnidirectional and another is cardioid, it is nothing like the full story about how a mic really performs. To get closer to the truth about how a mic performs in real life, you have to take measurements at a number of different frequencies. In Diagram R, these are represented by a blue line for 8kHz and a purple line for 16kHz.

    The first thing I’m sure you’ll notice is that these new measurements are nothing like as regular as those snappy graphics we got at 1kHz. Looking at the omnidirectional example on the left, the performance isn’t too bad, with -10dB at a couple of points by the time we get to 16kHz. Most omnis will exhibit something like this, and the most likely culprit is the way the capsule is physically mounted, the mass of the casing etc. We might use a microphone like this to record a piano in a concert hall and we would not have to sorry that reflected sound from the room was colored unduly by the mic.

    The plot on the right tells a different story, in that the curves at 8kHz and 16kHz are markedly different than the one at 1kHz. To better understand the problems this presents us with, you might want to go back to Diagram Q, which shows a typical frequency plot for a stage vocal mic (solid blue line). As I suggested in the previous post, that on-axis response is very usable.

    But look at the frequency response shown by the dotted blue line, which represents the same mic but 135° off axis. Would you want a mic curve like that? Well, bad luck fella, because the chances are you are already using one that performs like that, or even worse. I’ve already flagged this up in previous the thread I started on venue acoustics but I’ll say it again here: the off-axis frequency responses shown in Diagrams Q & R are typical of some of the best vocal stage mics out there. The inherent limitations of directional microphones have to be worked round and cannot be solved simply by throwing more money at the problem.

    Screen Shot 2019-08-10 at 15.57.04.png
     
  19. simoncroft

    simoncroft Tele-Meister

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    NB – It is worth pointing out that cheaper stage mics will not even achieve this level of performance. Those are the mics that usually come with only a single idealized plot, based on a single frequency. While it is not impossible to get good results from cheaper mics, you will need more skill and ingenuity. At the other end of the scale, really up-market mics often come with a split polar plot, showing maybe two frequencies per side. This is simply to make the plots for each frequency easier to read. The guys who make those mics aren’t trying to hide anything – they’re showing off.

    If the off-axis response of cardioid mics is so irregular, what can be done to minimize the coloration this will cause? The single most successful solution is to keep on-stage volume levels in check, so that mics are not picking up unintended sounds off-axis. It’s worth considering that in many situations, a loud on-stage amp will be picked up by multiple mics, all at different degrees off axis and all at different distances from the sound source. The affect on sound quality can only be negative.

    It also helps to use close miking techniques, but this can cause problems of its own if done to excess. For instance, two microphones used as a ‘stereo pair’ a few feet above a grand piano can give a balanced impression of the instrument. The closer the microphones are placed, the more they will focus on particular ranges of the instrument. In theory, you could address the problem by adding more and more mics, but the result is likely to sound less and less like the instrument you are attempting to amplify.

    Similarly, if you get in closer to brass instruments than about a yard (1m), you are no longer picking up the sound these instruments make in a room, due to the way in which the lower frequencies radiate from the bell. A similar situation exists with brasswind (woodwind). There is a further sting in the tail, which I’ll explain in a minute.

    Another approach is to use more directional microphones. As mentioned earlier, some cardioids are ‘tighter’ than others. There is also a ‘hypercardioid’ pattern that picks up even less from the sides than a cardioid but it’s not a free lunch. The tighter the pattern gets at the sides, the more pronounced the ‘lobe’ around 180° becomes. (You can see this on a cardioid as well, if you look again at Diagram R.) So, although a hypercardioid mic can reduce spill, it can also increase feedback problems if there is a monitor wedge facing directly at the performer.

    Now then, if you’ve ever cupped a hand over a stage mic, you’ll know how easily that simple action can push a mic into feedback. That’s because you’ve modified the polar response of the mic by mechanical means. Imagine what happens if you cup an entire trombone bell over a mic! Yep, and if you are using a hypercardioid mic, that lobe gets so much bigger as that bell looms over the mic…
     
  20. simoncroft

    simoncroft Tele-Meister

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    This is an appropriate point to introduce the concept of SPL (Sound Pressure Level). As well as being a useful way of describing the sound level from a rig, or the volume from any sound source, its also important when deciding if a particular mic is ‘up to the job’.

    If you’ve ever had your head in a bass drum when the beater connects with the head, you probably didn’t emerge muttering: “I reckon you’ve just unleashed anything in the region of 140dB on my unprotected ears. How interesting.” Similarly, if you’re walking towards a trumpet player, it’s likely you’ll think “gosh, that’s loud”, rather than: “When I was four yards away the SPL was in the 96dB region but now I’m only a foot away it must be in excess 130dB.” Far more likely you shouted: “Hey fella! You could turn me deaf doing that!”

    Well, mics have a point where they can either be damaged by the incoming SPL, or start to distort. Hence for percussion and brass especially, it is very useful to know whether the mic you intend to use can withstand the kind of SPL you are about to inflict on it.

    Next time, I’ll introduce some typical microphone design types (dynamics and condensers) and see how they compare in terms of SPL, sensitivity and frequency range – all of which are inter-related.
     
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