Showing posts with label Gain Structure. Show all posts
Showing posts with label Gain Structure. Show all posts
Thursday, February 9, 2012
Friday, February 3, 2012
Understanding Gain Structure :: TGP Webzine
Understanding Gain Structure :: TGP Webzine
Understanding Gain Structure
Tone Secrets Without Screwdrivers: Finding Mountains Of Tone In
A Guitar-Amp Rig by Understanding Gain Structure.
(Part one…or how I learned to stop worrying and love the master volume)
By Ray Parrish
Master volume is the work of the devil.....
…unless you know what master volume actually is…..unless you know that gain is not dirt…..distortion isn't just dirt…..saturation is only part of distortion…..that distortion means the signal coming out doesn't look like the signal that went in…..that distortion you don’t like is noise…..and distortion you do like is signal.
Volume and master volume aren't just about volume. They’re really where your tone lives. Most important of all, once it leaves your fingers, it’s all about the signal. What if I just want to find all the cool tones in my amp without ever using a screwdriver to open it up? What if I don’t want to build the amp……I just want to play it? Willie Johnson, I hear ya’!
No doubt there’s tons of very cool stuff hiding in any guitar/amp rig, some of it very hard to find for all but the initiated, so IMHO it really helps to know a little bit about how your rig thinks in order to find it all. By learning how amps and guitars think in general, a lot of those knobs on the front and rear panels will make gobs more sense and will greatly increase your power to sculpt tone before you ever touch an effect. So this article is the first in a series designed to show you, sans screwdriver, how to do just that with gain structure.
For starters, you need to define gain if you're going to control for it as an amp user. Very simply, gain is how much the signal strength grows by having passed through an amplifying device like a tube or transistor. Gain is not, however the same as distortion, saturation, drive, or any other popular industry misnomers. Gain is just a small signal made bigger. It follows that each place you take the signal and amplify it is a gain stage. And each one of your amplifying tubes or transistors is, in reality, (at least) one gain stage. A guitar amplifier is made up of several such gain stages, not just one. Controlling the gain you get from each gain stage, and what it does to all those that follow, is the secret of taking control of your tone.
The takeaway here is that these are all different phenomena, and the names in proper use are not interchangeable. The trouble is, in the gear biz, we do just that all the time, especially when it comes to putting decals on control panels. But, for the intelligent knob twister, knowing the difference is what puts you in command of your sound….in the front seat instead of the back seat. So literally speaking, what can you get from any given gain stage once you have control of it? What do you expect to hear when you twist a knob? If you’re going to layer these things, one gain stage after another, it helps to know what they are.
You probably know that if you push any gain stage hard enough, it will distort simply because it runs out of capacity to accurately amplify. Distortion is any sound that wasn’t in the original signal when it went in, and it can take many forms, including noise, clipping (aka dirt), altered frequency response, additional harmonics, or a variety of other more subtle and musical effects like sag and swirl. (we’ll dig into more technical—and useful--definitions in later articles. We’re good for now.)
Further, the various types of distortion come on in a kind of a little procession, one effect at a time, more or less. Typically tubes start out by adding the harmonics that make your guitar sound big and fill out your power chords. As you turn up, the rest of the circuit gets into the act and the EQ migrates toward the mids that cut through a mix, tailing off the muddy lows and screechy-loud highs. Approaching saturation, (but still what we think of as “clean”) the signal begins to compress, giving you that soaring lead sustain, not to mention filling out the body of your notes and voicing the harmonics and EQ shift ever more clearly. Further still, you get into the actual clipping that we call dirt. At extremes, the amp tone begins to dominate the heavily clipped guitar tone until finally it’s all about the amp, with the guitar relegated to defining the notes you’re playing and the voicing of your pick. It’s also here that complex interactions like swirl, sag, and bias excursion live, all of which do cool things if you know how to use them.
We’ll explore each of these behaviors more as need be, but for what we’re about to do here, this is more than enough to inform our inventive urges on a rainy day. Now that we know a few types of distortion, and understand that they come into play in a relatively gradual and progressive onset, let’s think about how we can control all those behaviors from the front panel. To know that, here’s a strange one that ya’ really gotta know: what does a knob actually do?
Controls of a Deluxe Reverb II
Oddly enough, when you get right down to the physics, knobs can only do one thing: attenuate. Even “active” knobs cannot add anything all by themselves—that takes a gain stage with—you guessed it—an attenuator. In an amp, gain, volume, tone, and master volume knobs simply control the attenuation of the output of each gain stage as the signal gets passed along. Yup, all three are known as attenuators, and all three do the same thing--attenuate. Weird! The first time I got my mitts around that concept, it really re-wired the way I thought about what I was doing to my gear.
By the way, for the time being we’re going to refer to knobs as attenuators, just to help picture what they’re really doing. We’ll talk about the different ways to wire attenuators later, (aka pots and all that) when we have a reason to care. For now we don’t. Right now, we’re actually going to draw a picture of the gain stages in an amp and show which attenuator controls which stage—in other words, we’re making that treasure map.
With complex amps, it helps to draw yourself a block diagram so you can see where each gain stage is, and where each volume-type attenuator (aka knob) is in the circuit path. Color me dumb, but I just can’t see electricity without a picture. It's also extremely handy to note where your tone knobs are, since they're actually frequency-specific volume attenuators too. So, just for instance, here's a block diagram of my knob-infested Deluxe Reverb II, a rather complex little chameleon of a beast, and one not particularly inclined to suffer fools...
Part 2
Understanding Gain Structure
Tone Secrets Without Screwdrivers: Finding Mountains Of Tone In
A Guitar-Amp Rig by Understanding Gain Structure.
(Part one…or how I learned to stop worrying and love the master volume)
By Ray Parrish
Master volume is the work of the devil.....
…unless you know what master volume actually is…..unless you know that gain is not dirt…..distortion isn't just dirt…..saturation is only part of distortion…..that distortion means the signal coming out doesn't look like the signal that went in…..that distortion you don’t like is noise…..and distortion you do like is signal.
Volume and master volume aren't just about volume. They’re really where your tone lives. Most important of all, once it leaves your fingers, it’s all about the signal. What if I just want to find all the cool tones in my amp without ever using a screwdriver to open it up? What if I don’t want to build the amp……I just want to play it? Willie Johnson, I hear ya’!
No doubt there’s tons of very cool stuff hiding in any guitar/amp rig, some of it very hard to find for all but the initiated, so IMHO it really helps to know a little bit about how your rig thinks in order to find it all. By learning how amps and guitars think in general, a lot of those knobs on the front and rear panels will make gobs more sense and will greatly increase your power to sculpt tone before you ever touch an effect. So this article is the first in a series designed to show you, sans screwdriver, how to do just that with gain structure.
For starters, you need to define gain if you're going to control for it as an amp user. Very simply, gain is how much the signal strength grows by having passed through an amplifying device like a tube or transistor. Gain is not, however the same as distortion, saturation, drive, or any other popular industry misnomers. Gain is just a small signal made bigger. It follows that each place you take the signal and amplify it is a gain stage. And each one of your amplifying tubes or transistors is, in reality, (at least) one gain stage. A guitar amplifier is made up of several such gain stages, not just one. Controlling the gain you get from each gain stage, and what it does to all those that follow, is the secret of taking control of your tone.
The takeaway here is that these are all different phenomena, and the names in proper use are not interchangeable. The trouble is, in the gear biz, we do just that all the time, especially when it comes to putting decals on control panels. But, for the intelligent knob twister, knowing the difference is what puts you in command of your sound….in the front seat instead of the back seat. So literally speaking, what can you get from any given gain stage once you have control of it? What do you expect to hear when you twist a knob? If you’re going to layer these things, one gain stage after another, it helps to know what they are.
You probably know that if you push any gain stage hard enough, it will distort simply because it runs out of capacity to accurately amplify. Distortion is any sound that wasn’t in the original signal when it went in, and it can take many forms, including noise, clipping (aka dirt), altered frequency response, additional harmonics, or a variety of other more subtle and musical effects like sag and swirl. (we’ll dig into more technical—and useful--definitions in later articles. We’re good for now.)
Further, the various types of distortion come on in a kind of a little procession, one effect at a time, more or less. Typically tubes start out by adding the harmonics that make your guitar sound big and fill out your power chords. As you turn up, the rest of the circuit gets into the act and the EQ migrates toward the mids that cut through a mix, tailing off the muddy lows and screechy-loud highs. Approaching saturation, (but still what we think of as “clean”) the signal begins to compress, giving you that soaring lead sustain, not to mention filling out the body of your notes and voicing the harmonics and EQ shift ever more clearly. Further still, you get into the actual clipping that we call dirt. At extremes, the amp tone begins to dominate the heavily clipped guitar tone until finally it’s all about the amp, with the guitar relegated to defining the notes you’re playing and the voicing of your pick. It’s also here that complex interactions like swirl, sag, and bias excursion live, all of which do cool things if you know how to use them.
We’ll explore each of these behaviors more as need be, but for what we’re about to do here, this is more than enough to inform our inventive urges on a rainy day. Now that we know a few types of distortion, and understand that they come into play in a relatively gradual and progressive onset, let’s think about how we can control all those behaviors from the front panel. To know that, here’s a strange one that ya’ really gotta know: what does a knob actually do?
Controls of a Deluxe Reverb II
Oddly enough, when you get right down to the physics, knobs can only do one thing: attenuate. Even “active” knobs cannot add anything all by themselves—that takes a gain stage with—you guessed it—an attenuator. In an amp, gain, volume, tone, and master volume knobs simply control the attenuation of the output of each gain stage as the signal gets passed along. Yup, all three are known as attenuators, and all three do the same thing--attenuate. Weird! The first time I got my mitts around that concept, it really re-wired the way I thought about what I was doing to my gear.
By the way, for the time being we’re going to refer to knobs as attenuators, just to help picture what they’re really doing. We’ll talk about the different ways to wire attenuators later, (aka pots and all that) when we have a reason to care. For now we don’t. Right now, we’re actually going to draw a picture of the gain stages in an amp and show which attenuator controls which stage—in other words, we’re making that treasure map.
With complex amps, it helps to draw yourself a block diagram so you can see where each gain stage is, and where each volume-type attenuator (aka knob) is in the circuit path. Color me dumb, but I just can’t see electricity without a picture. It's also extremely handy to note where your tone knobs are, since they're actually frequency-specific volume attenuators too. So, just for instance, here's a block diagram of my knob-infested Deluxe Reverb II, a rather complex little chameleon of a beast, and one not particularly inclined to suffer fools...
Part 2
Ins & Outs Of Gain Structure_ SOS Magazine
Ins & Outs Of Gain Structure
THE INS AND OUTS OF GAIN STRUCTURE
Setting up your gear for low noise and minimum distortion needn't be a nightmare. MARTIN WALKER guides you through the process, and shows you how to stand tall, even without much headroom.
Most people understand that if they want to get the best audio performance out of their studio equipment, input levels from sound sources must be high enough to ensure that noise levels remain very low by comparison, but not so high that they overload the equipment and cause distortion. However, for the best results, this optimisation process must be carried out at each stage of the amplifying chain, and this is where the concept of gain 'structure' comes from -- the tweaks are carried out from the very first input, all the way along the signal path, right to the end of the chain, whether the signal is being recorded onto DAT, or emerging from a loudspeaker.
MAKING A START
"Those with golden ears say that solid-state amplifiers start to sound 'edgy' in the final few dBs before clipping sets in."
If you record with acoustic instruments, the first stage for you will be to ensure that the input gain of your mixer's microphone input (or one of the fashionable stand-alone mic preamps) is set correctly for the levels coming out of the mic itself. Most mixers, even tiny ones, provide PFL (Pre-Fade Listen), and this allows you to monitor the level of a particular mixer channel after any EQ, but before the main channel fader. This is extremely useful when you want to listen to any sound in isolation, and also for initially setting up the input gain controls. When you press the PFL button, the signal will normally also be routed to one of the mixer meters, so that you can see its level. Even without PFL facilities you can achieve the same thing by first pulling all the channel faders right down, setting the master faders to unity (0dB), and then raising each channel fader in turn to the 0dB position.
Once you have some typical signal levels going through the mixer, you adjust a channel's gain control until the meter is hovering around the 0dB level (for a reasonably steady signal), or a bit higher (+6dB or so) if there are a lot of transients in the signal, since its average level will then be somewhat lower. If you're dealing with closely miked drums, some input levels may be so high that even with minimum mixer input gain you still have too much signal level; in this case you may have to switch in a pad, or use a less sensitive mic. Plugging the mic into a line input is not recommended, since the impedance values will be wrong.
FURTHER READING
Setting the right DAT recording level: SOS January 1995.
Noise and how to avoid it: SOS May 1995.
A Concise Guide to Compression & Limiting: SOS April 1996.
The Mysteries of Metering: SOS May 1996.
Minimising Mixer and Effects Noise: SOS July 1996.
Most stereo mixer inputs, the ones often used for electronic instruments such as synths and samplers, only provide a switch labelled +4/-10, instead of a fully variable gain control, and the best position of this switch can be determined in exactly the same way, using the channel PFL button. In most cases, if you can turn up the output level of your synth or effects unit to maximum, so that the switch can be set to the less sensitive '+4' position, you're likely to get slightly lower noise overall. Once all your inputs have been set up in this way, the channel faders are then used, with starting positions somewhere near the 0dB mark, to mix everything so that the final combined levels again peak at about the +6/+9dB mark on your output meters.
GETTING TWEAKY
So far, so good -- I'm sure most of you know about the above techniques already (although it's surprising how often I spot peoples' mixer output meters with only a couple of LEDs twitching near the bottom, or flashing red at the top of the range). What's a little more confusing is where different manufacturers choose to place the first red LED in their meter displays, and why. Many mixers have green LEDs up to 0dB, amber from 0dB to +6dB, and red for +9 and +12dB (the highest indicator). The idea is that if you see very occasional flashes of the +9dB LED on peaks, you'll be OK, but if the second red LED flashes as well (+12dB) you're approaching the point of distortion. In fact, all mixer manufacturers will have designed in a bit more headroom than this. Headroom is exactly what its name implies -- a bit more space over the metered limit before things overload -- and is traditionally the difference between average level and clip point.
This is where we find the huge difference between analogue and digital circuitry. If your mixer meter does occasionally go a little 'over the top', the mixer itself is unlikely to sound distorted, but if your mixer is feeding a digital recorder you'll almost certainly have to do another take. In a mixer, typically there will be at least another 6dB of output level available above the top LED before amplifier clipping occurs. Most mixers standardise on an output level of +4dBu (1.23V RMS) when the meters read 0dB VU. So when the top LED is just lit at +12dB VU, the actual output level emerging from the sockets will be +16dBu (4.9 volts RMS). If you look at your mixer spec to see its output level, it will give a figure of something like +22dBu maximum (9.76 volts RMS). This is 6dB higher than the top LED on the meter, and the extra headroom should ensure that your signals always emerge cleanly.
I say 'should' assuming that an amplifier will sound perfect right up to the clipping point, but this isn't always the case. Those with golden ears say that solid-state (transistor or FET) amplifiers start to sound 'edgy' in the final few dBs before clipping sets in. If you have the impression that some of your gear doesn't sound quite as good as it should when you drive it close to the clip point, you may be right. Fortunately most modern gear is quiet enough to be calibrated to run at slightly lower levels, to give a cleaner sound. Effects units, however, which are often the noisiest devices in the studio, are sometimes temperamental about overload, even for a few milliseconds, and so benefit from special treatment (see 'The Effects of Noise' box).
DISTORTION: NICE OR NASTY?
When you send audio equipment a signal large enough to overload its circuitry, each device will respond in a different way. Many people do overload their equipment for creative reasons, because the signal emerges with a different sound, and much development work is going on to produce computer software plug-ins which mimic the 'softer' overload characteristics of tube circuitry. Whereas a guitar amp normally benefits in a musical way from being overdriven, few people enjoy the sound of digital overload, and this is because of the nature of the distortion produced. The singing sound of guitar overdrive tends to be predominantly second harmonic, and the human ear finds this fairly pleasurable. For a start, it's only an octave away from the input signal and therefore easy for the ear to 'attach' to the overall sound. As digital circuitry becomes overloaded, it neatly clips off the top of the waveform, generating lots of third-harmonic distortion (not as nice as second, but still passable), but also lots of higher harmonics as well, extending to very high frequencies. Although the human ear can only pick up second-harmonic distortion when it reaches around 0.5%, eighth-harmonic distortion at as low a percentage as 0.01% is audible to humans -- which could be part of the reason why valve amps with high measured values of THD (Total Harmonic Distortion) often sound far better than transistor amps with THD values of 0.01%.
DAT'S THE WAY TO DO IT
With digital recording it's absolutely vital to avoid any overload at all. To be honest, although the mixer line-up procedure already discussed will optimise the gain structure of your analogue electronics, once a digital recorder is involved, most people will religiously watch the meter on that like a hawk, rather than relying on a mixer's meters. This is because most digital recorders have a Margin indicator as well, which shows the highest peak level recorded since recording began, and which holds this value until the Margin Reset button is pressed. In addition, since digital electronics are so sensitive to overload, these meters normally have a much faster response than the meters on a mixer, and may therefore show different readings as well. So now that the mixer is lined up so well, to complete the chain you have to calibrate your DAT machine to your mixer.
THE EFFECTS OF NOISE
Probably the easiest way for most people to achieve quieter mixes is to optimise the gain structure of their effect sends and returns, since effects units do tend to be the noisiest items in many studios. Try to ensure that most aux sends end up at about the 7 to 8 position, since this is normally the optimum position as far as mixer noise is concerned. However, since it's the output noise from the effect that can prove troublesome you should try to drive it as hard as possible at the input end. Many effects units have a 'Clip' or 'Overload' LED that comes on 5 or 6dB below clip point, but different units tend to react differently -- some sound horrendous even if this is exceeded for a few milliseconds, while others are more tolerant. If you can increase the input level control on the effects unit by a few dB, the Return fader on your mixer can be reduced by the same amount, leaving effects levels identical, but with correspondingly lower noise levels. Once you've performed these tweaks you'll probably notice a big improvement in the most obvious places -- at the beginning and end of tracks.
The digital meter on the DAT is calibrated in a rather different way, with the top of the scale reading 0dB, rather than 0VU appearing about two-thirds of the way up the mixer meter. You will often see a special mark on a DAT meter on or about the -12dB position. If you set up a 1kHz line-up oscillator on your mixer and adjust its level to exactly 0VU on the mixer output meters, you can go into Record Monitor mode on your DAT machine, and then slowly increase its input level control until this -12dB mark is reached. (Due to the large gaps between calibration marks on a DAT, looking at the Margin readout is a far more accurate way to do this, as it normally changes in much smaller increments, such as 0.5dB.) At this point, where 0dB VU as indicated by the mixer is equal to -12dB relative to Full Scale on the DAT meter (0VU = -12dBFS), you've calibrated your DAT machine so that mixer meters just touch the top red +12dB VU LED as the DAT machine reaches 0dB.
This -12dB reference level is fine for those recordings where every level is extremely well behaved, such as MIDI or sample playback, but live instruments recording is rarely so predictable. Since you still have headroom on your mixer beyond its Full Scale reading, you could reduce the DAT reference level to -18dBFS with 0dB VU on the mixer, to allow for unexpected transients. Now that more 20-bit converters are appearing, even on budget equipment, noise levels are also dropping, and there is a school of thought which says that using a reference level such as -18dBFS doesn't compromise noise levels significantly, whatever the dynamic range of your music, and at least it lets you return to looking at your mixer meters, without having to worry so much about the odd extra dB ruining a digital recording.
One thing to watch out for here: you may come across digital recorders that try to mimic their analogue counterparts by setting their internal reference level to typical mixer output levels. When your mixer reads 0dB VU (normally emerging at +4dBu), an Alesis ADAT may still only be reading about -15dB on its own meter. Although this gives you plenty of headroom, to reach 0dB on the ADAT meter will need +19dBu output from the mixer, which is getting perilously close to the clipping point of many small mixers -- and, as already mentioned, your mixer may not sound quite so clean in the final few dBs before clipping. If you find this is a problem, you might try using the digital recorder at its -10dBV input sensitivity, which will let you 'go all the way' without risking output clipping of your mixer.
For both live and studio work it's also common to patch in a compressor at the mixer buss insert point, and in some cases an additional 'brick-wall' limiter set to a level just below the digital clip point, to ensure that nothing gets through to overload the digital side. High-end processors such as the TC Electronic Finaliser even include a fine level adjuster for the limiter, calibrated in 0.01dB increments below 0dBFS, and since many digital recorders don't have proper 'Over' indicators (see The Digital 'Over'), this ensures that you never get erroneous readings, since the record signal will never actually reach 0dB.
HARD OR SOFT?
Now that digital recording is so much a part of the modern studio, a completely mathematical approach can be adopted. Since the digital signals are simply a stream of '1's and '0's, gain can be adjusted by multiplying or dividing digital values, and this is equivalent to amplification or gain reduction in the analogue domain. However, digital processes have one big advantage -- by looking ahead in the waveform, compression/limiting algorithms can anticipate transients, rather than having to react to them as quickly as possible after they happen, as in the case of the analogue compressor.
NOISE AND DYNAMIC RANGE
There is still much confusion between signal-to-noise ratio and dynamic range, especially where digital signals are concerned. Signal-to-noise ratio is the RMS level of the noise with no signal applied, expressed in dB below maximum level. Dynamic range is defined as the ratio of the loudest (undistorted) signal to that of the quietest (discernible) signal in a unit or system, as expressed in decibels (dB). Dynamic range is often said to be a subjective judgement more than a measurement -- you can compare the dynamic range of two systems empirically with identical listening tests, by applying a 1kHz tone, and see how low you can make it before it is undetectable.
The maximum signal-to-noise ratio of a 16-bit recording is 96dB, since each bit contributes 6dB to the total. If you leave 6dB of headroom on your digital recording, to prevent any unexpected peaks from causing clipping, you immediately reduce this to 90dB. The background noise level will depend on the converters, as well as the design of the rest of the circuitry. Due to the confusions, even between manufacturers, on how these figures should be measured, Crystal Semiconductor (the well-known designers of A/D and D/A converter chips, as used by many companies worldwide) have suggested a standard method for the following measurement procedure. They define Dynamic range (DR) as the ratio of the full signal level to the RMS noise floor, in the presence of signal, expressed in dB FS. The addition of a low-level signal (a suggested 1kHz sinewave at a level of -60dB FS) ensures that any noise-gate circuitry is bypassed, but of course this signal must be notched out before the actual measurement is taken. The final figure quoted is also likely to be A-weighted, which takes account of the characteristics of the ear, which is more sensitive to frequencies between 2k(dot)z and 4kHz. 'A-weighted' figures tend to make comparing figures between different hardware components easier.
Normalisation is another gain adjustment, but this time it's carried out after recording, by bringing the maximum peak level to the maximum allowable digital level, to make sure that the signal is as 'hot' as possible. It does not increase the dynamic range of the programme material, since low-level signals will be brought up by exactly the same amount as high-level ones, and neither does it ensure that every track destined for an album will end up at the same perceived loudness, since this depends on average levels, and not peak ones. Many recordings will only have a few occasional peaks approaching 0dB, and the average level can nearly always be brought up by at least 3 or 4dB, simply by compressing these short transients, without having an obvious audible effect. You can do this with a limiter, or using the software approach of plug-ins like Waves' L1 Ultramaximiser, and Steinberg's Loudness Maximiser.
"Ultimately, being thorough in your approach to gain structure will ensure that you only hear distortion when it's part of your music, and that quiet passages remain free of unwanted background hisses and hums."
If you're trying to make your recordings sound as 'loud' as commercial releases, it's best to monitor all digital signals through one high-quality digital/analogue converter -- listen to your CDs, hard disk recordings, and so on, and then you can compare them directly.
Normalisation does ensure that on cheaper playback systems, where system background noise is more of a problem, your recordings will make use of the top end of the dynamic range. It should always be carried out as the final operation, after any other digital editing, since you're asking for trouble if you tweak the digital signal any more once it contains peaks at the maximum theoretical level. Also, since both of the above plug-ins can take advantage of noise-shaped dithering (for better low-level resolution), this can also raise levels, particularly at higher frequencies. For this reason, many people play safe, and normalise to a figure just below 0dB -- Waves recommend -0.3dB when using their L1 Ultramaximiser during mastering for CD.
THE FINAL TOUCHES
Ultimately, being thorough in your approach to gain structure will ensure that you only hear distortion when it's part of your music, and that quiet passages remain free of unwanted background hisses and hums. Noise gates and muting can remove all the background grunge once it falls beneath a threshold level, but with attention to detail and careful wiring (preferably with balanced lines) your music will sound more transparent even at normal levels if the noise floor is as low as possible. Sadly, digital artefacts often sound more objectionable than analogue ones, because rather than being random in nature (a steady background hiss that the ear tends to ignore if low in level), they tend to be tied into the signal itself, and are therefore more noticeable. At low levels, where the converters run out of resolution, smooth waveforms begin to resemble a staircase, which gives a gritty sound, known as quantisation noise.
THE DIGITAL 'OVER'
When dealing with digital recorders, any signal that flashes the Over indicator on the input level meter of the digital recorder will sound dreadful on playback -- there is no headroom with a digital meter. In fact, very few digital recorders actually have a proper digital 'over' indicator -- if you think about it, once the signal gets to 0dB, it just can't get any higher. Many so-called Over indicators are actually measuring analogue levels, so that they can indicate a level which exceeded the calibrated digital peak level. This is why you can overload the inputs of some digital machines and the tape you've just recorded never shows any overload indication on replay. Other machines which do flash an overload may well be reading 0dB and assuming that the signal might have overloaded.
The clever machines use a different method to determine whether or not a real overload has occurred. Just as the highest peak of a signal touches the 0dB mark, a single sample will be recorded with a value of 0dB. If the input level goes any higher, several samples in a row will be at this 0dB point, and this is likely to be because of overload. So some manufacturers count consecutive 0dB values -- the Over indicator on a Sony 1630 machine will indicate an overload if three samples in a row are detected at 0dB. However, at 44.1kHz three samples lasts only a few tens of microseconds, which is generally regarded as inaudible -- other manufacturers use four, five or six samples in a row. The beauty of the sample-counting Over indicator is that it will also work with a digital input, and pick up recordings that have previously been overloaded.
The last year has seen many more affordable digital converters appearing in mid-price to budget equipment, and these have lower noise floors than equivalently priced equipment that is a couple of years old. Now, more than ever, it's worth giving your studio the once-over to optimise everything in the audio chain.
THE INS AND OUTS OF GAIN STRUCTURE
Setting up your gear for low noise and minimum distortion needn't be a nightmare. MARTIN WALKER guides you through the process, and shows you how to stand tall, even without much headroom.
Most people understand that if they want to get the best audio performance out of their studio equipment, input levels from sound sources must be high enough to ensure that noise levels remain very low by comparison, but not so high that they overload the equipment and cause distortion. However, for the best results, this optimisation process must be carried out at each stage of the amplifying chain, and this is where the concept of gain 'structure' comes from -- the tweaks are carried out from the very first input, all the way along the signal path, right to the end of the chain, whether the signal is being recorded onto DAT, or emerging from a loudspeaker.
MAKING A START
"Those with golden ears say that solid-state amplifiers start to sound 'edgy' in the final few dBs before clipping sets in."
If you record with acoustic instruments, the first stage for you will be to ensure that the input gain of your mixer's microphone input (or one of the fashionable stand-alone mic preamps) is set correctly for the levels coming out of the mic itself. Most mixers, even tiny ones, provide PFL (Pre-Fade Listen), and this allows you to monitor the level of a particular mixer channel after any EQ, but before the main channel fader. This is extremely useful when you want to listen to any sound in isolation, and also for initially setting up the input gain controls. When you press the PFL button, the signal will normally also be routed to one of the mixer meters, so that you can see its level. Even without PFL facilities you can achieve the same thing by first pulling all the channel faders right down, setting the master faders to unity (0dB), and then raising each channel fader in turn to the 0dB position.
Once you have some typical signal levels going through the mixer, you adjust a channel's gain control until the meter is hovering around the 0dB level (for a reasonably steady signal), or a bit higher (+6dB or so) if there are a lot of transients in the signal, since its average level will then be somewhat lower. If you're dealing with closely miked drums, some input levels may be so high that even with minimum mixer input gain you still have too much signal level; in this case you may have to switch in a pad, or use a less sensitive mic. Plugging the mic into a line input is not recommended, since the impedance values will be wrong.
FURTHER READING
Setting the right DAT recording level: SOS January 1995.
Noise and how to avoid it: SOS May 1995.
A Concise Guide to Compression & Limiting: SOS April 1996.
The Mysteries of Metering: SOS May 1996.
Minimising Mixer and Effects Noise: SOS July 1996.
Most stereo mixer inputs, the ones often used for electronic instruments such as synths and samplers, only provide a switch labelled +4/-10, instead of a fully variable gain control, and the best position of this switch can be determined in exactly the same way, using the channel PFL button. In most cases, if you can turn up the output level of your synth or effects unit to maximum, so that the switch can be set to the less sensitive '+4' position, you're likely to get slightly lower noise overall. Once all your inputs have been set up in this way, the channel faders are then used, with starting positions somewhere near the 0dB mark, to mix everything so that the final combined levels again peak at about the +6/+9dB mark on your output meters.
GETTING TWEAKY
So far, so good -- I'm sure most of you know about the above techniques already (although it's surprising how often I spot peoples' mixer output meters with only a couple of LEDs twitching near the bottom, or flashing red at the top of the range). What's a little more confusing is where different manufacturers choose to place the first red LED in their meter displays, and why. Many mixers have green LEDs up to 0dB, amber from 0dB to +6dB, and red for +9 and +12dB (the highest indicator). The idea is that if you see very occasional flashes of the +9dB LED on peaks, you'll be OK, but if the second red LED flashes as well (+12dB) you're approaching the point of distortion. In fact, all mixer manufacturers will have designed in a bit more headroom than this. Headroom is exactly what its name implies -- a bit more space over the metered limit before things overload -- and is traditionally the difference between average level and clip point.
This is where we find the huge difference between analogue and digital circuitry. If your mixer meter does occasionally go a little 'over the top', the mixer itself is unlikely to sound distorted, but if your mixer is feeding a digital recorder you'll almost certainly have to do another take. In a mixer, typically there will be at least another 6dB of output level available above the top LED before amplifier clipping occurs. Most mixers standardise on an output level of +4dBu (1.23V RMS) when the meters read 0dB VU. So when the top LED is just lit at +12dB VU, the actual output level emerging from the sockets will be +16dBu (4.9 volts RMS). If you look at your mixer spec to see its output level, it will give a figure of something like +22dBu maximum (9.76 volts RMS). This is 6dB higher than the top LED on the meter, and the extra headroom should ensure that your signals always emerge cleanly.
I say 'should' assuming that an amplifier will sound perfect right up to the clipping point, but this isn't always the case. Those with golden ears say that solid-state (transistor or FET) amplifiers start to sound 'edgy' in the final few dBs before clipping sets in. If you have the impression that some of your gear doesn't sound quite as good as it should when you drive it close to the clip point, you may be right. Fortunately most modern gear is quiet enough to be calibrated to run at slightly lower levels, to give a cleaner sound. Effects units, however, which are often the noisiest devices in the studio, are sometimes temperamental about overload, even for a few milliseconds, and so benefit from special treatment (see 'The Effects of Noise' box).
DISTORTION: NICE OR NASTY?
When you send audio equipment a signal large enough to overload its circuitry, each device will respond in a different way. Many people do overload their equipment for creative reasons, because the signal emerges with a different sound, and much development work is going on to produce computer software plug-ins which mimic the 'softer' overload characteristics of tube circuitry. Whereas a guitar amp normally benefits in a musical way from being overdriven, few people enjoy the sound of digital overload, and this is because of the nature of the distortion produced. The singing sound of guitar overdrive tends to be predominantly second harmonic, and the human ear finds this fairly pleasurable. For a start, it's only an octave away from the input signal and therefore easy for the ear to 'attach' to the overall sound. As digital circuitry becomes overloaded, it neatly clips off the top of the waveform, generating lots of third-harmonic distortion (not as nice as second, but still passable), but also lots of higher harmonics as well, extending to very high frequencies. Although the human ear can only pick up second-harmonic distortion when it reaches around 0.5%, eighth-harmonic distortion at as low a percentage as 0.01% is audible to humans -- which could be part of the reason why valve amps with high measured values of THD (Total Harmonic Distortion) often sound far better than transistor amps with THD values of 0.01%.
DAT'S THE WAY TO DO IT
With digital recording it's absolutely vital to avoid any overload at all. To be honest, although the mixer line-up procedure already discussed will optimise the gain structure of your analogue electronics, once a digital recorder is involved, most people will religiously watch the meter on that like a hawk, rather than relying on a mixer's meters. This is because most digital recorders have a Margin indicator as well, which shows the highest peak level recorded since recording began, and which holds this value until the Margin Reset button is pressed. In addition, since digital electronics are so sensitive to overload, these meters normally have a much faster response than the meters on a mixer, and may therefore show different readings as well. So now that the mixer is lined up so well, to complete the chain you have to calibrate your DAT machine to your mixer.
THE EFFECTS OF NOISE
Probably the easiest way for most people to achieve quieter mixes is to optimise the gain structure of their effect sends and returns, since effects units do tend to be the noisiest items in many studios. Try to ensure that most aux sends end up at about the 7 to 8 position, since this is normally the optimum position as far as mixer noise is concerned. However, since it's the output noise from the effect that can prove troublesome you should try to drive it as hard as possible at the input end. Many effects units have a 'Clip' or 'Overload' LED that comes on 5 or 6dB below clip point, but different units tend to react differently -- some sound horrendous even if this is exceeded for a few milliseconds, while others are more tolerant. If you can increase the input level control on the effects unit by a few dB, the Return fader on your mixer can be reduced by the same amount, leaving effects levels identical, but with correspondingly lower noise levels. Once you've performed these tweaks you'll probably notice a big improvement in the most obvious places -- at the beginning and end of tracks.
The digital meter on the DAT is calibrated in a rather different way, with the top of the scale reading 0dB, rather than 0VU appearing about two-thirds of the way up the mixer meter. You will often see a special mark on a DAT meter on or about the -12dB position. If you set up a 1kHz line-up oscillator on your mixer and adjust its level to exactly 0VU on the mixer output meters, you can go into Record Monitor mode on your DAT machine, and then slowly increase its input level control until this -12dB mark is reached. (Due to the large gaps between calibration marks on a DAT, looking at the Margin readout is a far more accurate way to do this, as it normally changes in much smaller increments, such as 0.5dB.) At this point, where 0dB VU as indicated by the mixer is equal to -12dB relative to Full Scale on the DAT meter (0VU = -12dBFS), you've calibrated your DAT machine so that mixer meters just touch the top red +12dB VU LED as the DAT machine reaches 0dB.
This -12dB reference level is fine for those recordings where every level is extremely well behaved, such as MIDI or sample playback, but live instruments recording is rarely so predictable. Since you still have headroom on your mixer beyond its Full Scale reading, you could reduce the DAT reference level to -18dBFS with 0dB VU on the mixer, to allow for unexpected transients. Now that more 20-bit converters are appearing, even on budget equipment, noise levels are also dropping, and there is a school of thought which says that using a reference level such as -18dBFS doesn't compromise noise levels significantly, whatever the dynamic range of your music, and at least it lets you return to looking at your mixer meters, without having to worry so much about the odd extra dB ruining a digital recording.
One thing to watch out for here: you may come across digital recorders that try to mimic their analogue counterparts by setting their internal reference level to typical mixer output levels. When your mixer reads 0dB VU (normally emerging at +4dBu), an Alesis ADAT may still only be reading about -15dB on its own meter. Although this gives you plenty of headroom, to reach 0dB on the ADAT meter will need +19dBu output from the mixer, which is getting perilously close to the clipping point of many small mixers -- and, as already mentioned, your mixer may not sound quite so clean in the final few dBs before clipping. If you find this is a problem, you might try using the digital recorder at its -10dBV input sensitivity, which will let you 'go all the way' without risking output clipping of your mixer.
For both live and studio work it's also common to patch in a compressor at the mixer buss insert point, and in some cases an additional 'brick-wall' limiter set to a level just below the digital clip point, to ensure that nothing gets through to overload the digital side. High-end processors such as the TC Electronic Finaliser even include a fine level adjuster for the limiter, calibrated in 0.01dB increments below 0dBFS, and since many digital recorders don't have proper 'Over' indicators (see The Digital 'Over'), this ensures that you never get erroneous readings, since the record signal will never actually reach 0dB.
HARD OR SOFT?
Now that digital recording is so much a part of the modern studio, a completely mathematical approach can be adopted. Since the digital signals are simply a stream of '1's and '0's, gain can be adjusted by multiplying or dividing digital values, and this is equivalent to amplification or gain reduction in the analogue domain. However, digital processes have one big advantage -- by looking ahead in the waveform, compression/limiting algorithms can anticipate transients, rather than having to react to them as quickly as possible after they happen, as in the case of the analogue compressor.
NOISE AND DYNAMIC RANGE
There is still much confusion between signal-to-noise ratio and dynamic range, especially where digital signals are concerned. Signal-to-noise ratio is the RMS level of the noise with no signal applied, expressed in dB below maximum level. Dynamic range is defined as the ratio of the loudest (undistorted) signal to that of the quietest (discernible) signal in a unit or system, as expressed in decibels (dB). Dynamic range is often said to be a subjective judgement more than a measurement -- you can compare the dynamic range of two systems empirically with identical listening tests, by applying a 1kHz tone, and see how low you can make it before it is undetectable.
The maximum signal-to-noise ratio of a 16-bit recording is 96dB, since each bit contributes 6dB to the total. If you leave 6dB of headroom on your digital recording, to prevent any unexpected peaks from causing clipping, you immediately reduce this to 90dB. The background noise level will depend on the converters, as well as the design of the rest of the circuitry. Due to the confusions, even between manufacturers, on how these figures should be measured, Crystal Semiconductor (the well-known designers of A/D and D/A converter chips, as used by many companies worldwide) have suggested a standard method for the following measurement procedure. They define Dynamic range (DR) as the ratio of the full signal level to the RMS noise floor, in the presence of signal, expressed in dB FS. The addition of a low-level signal (a suggested 1kHz sinewave at a level of -60dB FS) ensures that any noise-gate circuitry is bypassed, but of course this signal must be notched out before the actual measurement is taken. The final figure quoted is also likely to be A-weighted, which takes account of the characteristics of the ear, which is more sensitive to frequencies between 2k(dot)z and 4kHz. 'A-weighted' figures tend to make comparing figures between different hardware components easier.
Normalisation is another gain adjustment, but this time it's carried out after recording, by bringing the maximum peak level to the maximum allowable digital level, to make sure that the signal is as 'hot' as possible. It does not increase the dynamic range of the programme material, since low-level signals will be brought up by exactly the same amount as high-level ones, and neither does it ensure that every track destined for an album will end up at the same perceived loudness, since this depends on average levels, and not peak ones. Many recordings will only have a few occasional peaks approaching 0dB, and the average level can nearly always be brought up by at least 3 or 4dB, simply by compressing these short transients, without having an obvious audible effect. You can do this with a limiter, or using the software approach of plug-ins like Waves' L1 Ultramaximiser, and Steinberg's Loudness Maximiser.
"Ultimately, being thorough in your approach to gain structure will ensure that you only hear distortion when it's part of your music, and that quiet passages remain free of unwanted background hisses and hums."
If you're trying to make your recordings sound as 'loud' as commercial releases, it's best to monitor all digital signals through one high-quality digital/analogue converter -- listen to your CDs, hard disk recordings, and so on, and then you can compare them directly.
Normalisation does ensure that on cheaper playback systems, where system background noise is more of a problem, your recordings will make use of the top end of the dynamic range. It should always be carried out as the final operation, after any other digital editing, since you're asking for trouble if you tweak the digital signal any more once it contains peaks at the maximum theoretical level. Also, since both of the above plug-ins can take advantage of noise-shaped dithering (for better low-level resolution), this can also raise levels, particularly at higher frequencies. For this reason, many people play safe, and normalise to a figure just below 0dB -- Waves recommend -0.3dB when using their L1 Ultramaximiser during mastering for CD.
THE FINAL TOUCHES
Ultimately, being thorough in your approach to gain structure will ensure that you only hear distortion when it's part of your music, and that quiet passages remain free of unwanted background hisses and hums. Noise gates and muting can remove all the background grunge once it falls beneath a threshold level, but with attention to detail and careful wiring (preferably with balanced lines) your music will sound more transparent even at normal levels if the noise floor is as low as possible. Sadly, digital artefacts often sound more objectionable than analogue ones, because rather than being random in nature (a steady background hiss that the ear tends to ignore if low in level), they tend to be tied into the signal itself, and are therefore more noticeable. At low levels, where the converters run out of resolution, smooth waveforms begin to resemble a staircase, which gives a gritty sound, known as quantisation noise.
THE DIGITAL 'OVER'
When dealing with digital recorders, any signal that flashes the Over indicator on the input level meter of the digital recorder will sound dreadful on playback -- there is no headroom with a digital meter. In fact, very few digital recorders actually have a proper digital 'over' indicator -- if you think about it, once the signal gets to 0dB, it just can't get any higher. Many so-called Over indicators are actually measuring analogue levels, so that they can indicate a level which exceeded the calibrated digital peak level. This is why you can overload the inputs of some digital machines and the tape you've just recorded never shows any overload indication on replay. Other machines which do flash an overload may well be reading 0dB and assuming that the signal might have overloaded.
The clever machines use a different method to determine whether or not a real overload has occurred. Just as the highest peak of a signal touches the 0dB mark, a single sample will be recorded with a value of 0dB. If the input level goes any higher, several samples in a row will be at this 0dB point, and this is likely to be because of overload. So some manufacturers count consecutive 0dB values -- the Over indicator on a Sony 1630 machine will indicate an overload if three samples in a row are detected at 0dB. However, at 44.1kHz three samples lasts only a few tens of microseconds, which is generally regarded as inaudible -- other manufacturers use four, five or six samples in a row. The beauty of the sample-counting Over indicator is that it will also work with a digital input, and pick up recordings that have previously been overloaded.
The last year has seen many more affordable digital converters appearing in mid-price to budget equipment, and these have lower noise floors than equivalently priced equipment that is a couple of years old. Now, more than ever, it's worth giving your studio the once-over to optimise everything in the audio chain.
The importance of gain structure when recording and mixing music. - MTT Open - Music Think Tank
The importance of gain structure when recording and mixing music. - MTT Open - Music Think Tank
The importance of gain structure when recording and mixing music.
Tweet Barry Gardner | Print Article | Post a Comment
As a musician it is highly recommended that you use 24 bit resolution in your digital audio workstation. This affords a number of real advantages and not just when processing the audio with plug in software. When you set your DAW to 24 bit you have allowed yourself to record at a much lower level without any technical detriment. The theoretical noise floor at 24 bit is significantly lower than that of a 16bit recording. This means that you can now record signals that peak at around -18dBFS. Thats sounds low but in fact this is equivalent to the electrical level that would have been understood as nominal in a large NEVE or SSL console i.e. 0Vu. In a digital system -18dBFS is referenced to +4dBu (1.23 volts), the same can be said of 0Vu. So there is no need to record at high recording levels when using 24 resolution. I think the confusion may have crept in for 2 reasons, we we recommended that hot signals were good at 16 bit and also the saying “hit zero” may have worked it’s way into the minds of musicians as a hang over from the days of large consoles and Vu metering.
Now we can consider mixing headroom on your master output bus. As with recording at 24 bit there is no need to peak so high on the outputs. Simply use a “peaky” signal to reference against. When you start you mix use a kick drum or snare as a reference to built the mix around. Peak the signal at -18dBFS on your stereo output meter. Once you have started mixing other instruments try not to adjust the level of this reference and build the mix around it. This will have built in plenty of headroom and you should not have any issues with peaking above 0dBFS.
Another bonus is going to be cleaner monitoring, this is because the small electronic devices in your sound card will no longer be operating as close to their technical design limits. This means less distortion should be present in the signal monitored. Low cost, aged design op amps have a habit of increasing their THD as they get closer to their maximum amplification capabilities. The only thing you need to do when working at -18dBFS is to raise your monitoring volume on your amplifier or monitor controller to compensate for the lower operating level.
Many pople will put a limiter on the master bus to keep the levels up and at the same time from going over digital zero. However, this is an extreme dynamic process and will quite possibly negatively influence your mix decisions. Your drum peaks will be arrested in their prime. This is where the punch of the mix resides. Instead if you require a limiter later in self finalizing you can export your mix at 24 bit and limit in a new project on the stereo file. Self finalizing or professional online mastering is best done in a separate project with fresh ears. This lower operating level has many benefits for musicians and is a great way to end up with better, cleaner sounding mixes.
The importance of gain structure when recording and mixing music.
Tweet Barry Gardner | Print Article | Post a Comment
As a musician it is highly recommended that you use 24 bit resolution in your digital audio workstation. This affords a number of real advantages and not just when processing the audio with plug in software. When you set your DAW to 24 bit you have allowed yourself to record at a much lower level without any technical detriment. The theoretical noise floor at 24 bit is significantly lower than that of a 16bit recording. This means that you can now record signals that peak at around -18dBFS. Thats sounds low but in fact this is equivalent to the electrical level that would have been understood as nominal in a large NEVE or SSL console i.e. 0Vu. In a digital system -18dBFS is referenced to +4dBu (1.23 volts), the same can be said of 0Vu. So there is no need to record at high recording levels when using 24 resolution. I think the confusion may have crept in for 2 reasons, we we recommended that hot signals were good at 16 bit and also the saying “hit zero” may have worked it’s way into the minds of musicians as a hang over from the days of large consoles and Vu metering.
Now we can consider mixing headroom on your master output bus. As with recording at 24 bit there is no need to peak so high on the outputs. Simply use a “peaky” signal to reference against. When you start you mix use a kick drum or snare as a reference to built the mix around. Peak the signal at -18dBFS on your stereo output meter. Once you have started mixing other instruments try not to adjust the level of this reference and build the mix around it. This will have built in plenty of headroom and you should not have any issues with peaking above 0dBFS.
Another bonus is going to be cleaner monitoring, this is because the small electronic devices in your sound card will no longer be operating as close to their technical design limits. This means less distortion should be present in the signal monitored. Low cost, aged design op amps have a habit of increasing their THD as they get closer to their maximum amplification capabilities. The only thing you need to do when working at -18dBFS is to raise your monitoring volume on your amplifier or monitor controller to compensate for the lower operating level.
Many pople will put a limiter on the master bus to keep the levels up and at the same time from going over digital zero. However, this is an extreme dynamic process and will quite possibly negatively influence your mix decisions. Your drum peaks will be arrested in their prime. This is where the punch of the mix resides. Instead if you require a limiter later in self finalizing you can export your mix at 24 bit and limit in a new project on the stereo file. Self finalizing or professional online mastering is best done in a separate project with fresh ears. This lower operating level has many benefits for musicians and is a great way to end up with better, cleaner sounding mixes.
Live Sound: The Basics Of Gain Structure - Pro Sound Web
Live Sound: The Basics Of Gain Structure - Pro Sound Web
The Basics Of Gain Structure
General guidelines to setting the gain structure for an audio system and making final adjustments.
December 02, 2010, by Chuck McGregor
Although this is not a rigorous treatment of the subject, the following are general guidelines that may be helpful in setting the gain structure for an audio system and making final adjustments to it for best results.
A Basic Procedure
As a minimum, you need to use at least a voltmeter along with the maximum output voltage specification for each piece of electronics.
The preferable tool is an oscilloscope that can be used to observe the signal directly.
If neither of these are available, you can use the level or clipping indicators on each piece of equipment in the signal chain.
Without some method of determining the clipping point for each piece of equipment, you cannot expect to optimize the gain structure.
Before setting gain structure disconnect the loudspeaker(s) from the amplifiers. If controllers are used that sense the amplifier outputs do not disconnect the sense lines.
The basic procedure is to use a test signal (a sine wave signal is ideal) and set the first piece of equipment in the signal chain (usually the mixer) so it is just below maximum voltage output as read on the voltmeter or on equipment’s output meter or just below clipping as observed on the oscilloscope.
Without changing this signal level adjust the level controls on each piece of equipment following, including the power amplifiers, so it is just below its maximum output.
You will find that the input level controls on the power amplifier will end up being set anywhere from 10 dB to over 20 dB of attenuation.
Note:
Due to differences in the capabilities of devices in the signal chain, it may not be possible to achieve the results exactly as stated.
Gain structure should be set after any equalization is set for the system so that any boosts (which reduce headroom in the equalizer) are taken into account.
For each device, make sure it is the output and not the input that is clipping.
Always make sure that the any limiters in the system go into limiting prior to anything else in the system going into clipping. In this way the limiter, rather than a clipped signal somewhere else in the signal chain, will control the system’s maximum output.
Remove the test signal, turn off all equipment and reconnect the loudspeaker(s) and the system is ready for level balancing, assuming you have more than one loudspeaker.
If you have only one loudspeaker the system should be ready for use.
Residual Noise
Horn loaded loudspeakers and drivers have high sensitivities which means that they produce a relatively high output for a given electrical input.
This includes the residual electronic noise of an audio system.
By setting gain structure properly and using high quality professional electronics, this noise should be at or near inaudibility.
Among the noisier electronic devices are 16 bit digital devices such as some signal delays.
They have signal to noise ratios that are only 90 dB. However, if the gain structure is set correctly this means that, for example, if the system can produce 120 dB SPL at maximum output, the residual noise should be about 30 dB SPL.
This would be acceptable for a quiet recording studio. If residual noise is a problem, gain structure is usually the culprit – it is never the loudspeakers.
Level Balancing
Once the system gain structure is set, the level balances can be adjusted. This may mean the levels between HF and LF sections of a biamplified loudspeaker, a full-range loudspeaker to subwoofer level, levels between multiple loudspeakers, or between main and delayed loudspeaker array.
The idea is to make the system sound the best it can without using any equalization. This may be done using acoustic test equipment such as an RTA (real-time analyzer), TEF analyzer or similar. You must always determine the final level balance by listening to a variety of known program material.
Level balancing can also be done entirely by ear if acoustic test equipment is not available. In any case, the preferred method of adjusting levels for balancing is to use the amplifier input level controls.
Important:
In order to maintain the system’s dynamic range that was maximized by setting the proper gain structure, do not turn up the input level controls of any of the amplifiers.
For example, if you decide that a subwoofer is not loud enough, do not turn up the input level control of its amplifier. Instead, turn down the input level of the full-range loudspeaker’s amplifier.
Equalization
Once level balances are set, you can then equalize the loudspeaker(s), if necessary. Community loudspeakers are optimized for highly accurate and well balanced reproduction “out-of-the-box”.
Generally, equalization should only be needed to eliminate difficult feedback frequencies or to adapt the system to a difficult acoustic environment.
You should not need more than a few dB of boost or cut equalization for any particular range of frequencies. The best equalization techniques involve cutting rather than boosting frequencies.
Final Results
Once the above procedures are followed, your loudspeaker should reproduce audio cleanly, clearly and with all frequencies in good balance.
Noise should not be audible and you should be able to drive the amplifier(s) in the system to maximum output on normal program material with no significant distortion or other undesirable sound. If limiters are used, the onset of limiting should occur just before any amplifier clipping.
The Basics Of Gain Structure
General guidelines to setting the gain structure for an audio system and making final adjustments.
December 02, 2010, by Chuck McGregor
Although this is not a rigorous treatment of the subject, the following are general guidelines that may be helpful in setting the gain structure for an audio system and making final adjustments to it for best results.
A Basic Procedure
As a minimum, you need to use at least a voltmeter along with the maximum output voltage specification for each piece of electronics.
The preferable tool is an oscilloscope that can be used to observe the signal directly.
If neither of these are available, you can use the level or clipping indicators on each piece of equipment in the signal chain.
Without some method of determining the clipping point for each piece of equipment, you cannot expect to optimize the gain structure.
Before setting gain structure disconnect the loudspeaker(s) from the amplifiers. If controllers are used that sense the amplifier outputs do not disconnect the sense lines.
The basic procedure is to use a test signal (a sine wave signal is ideal) and set the first piece of equipment in the signal chain (usually the mixer) so it is just below maximum voltage output as read on the voltmeter or on equipment’s output meter or just below clipping as observed on the oscilloscope.
Without changing this signal level adjust the level controls on each piece of equipment following, including the power amplifiers, so it is just below its maximum output.
You will find that the input level controls on the power amplifier will end up being set anywhere from 10 dB to over 20 dB of attenuation.
Note:
Due to differences in the capabilities of devices in the signal chain, it may not be possible to achieve the results exactly as stated.
Gain structure should be set after any equalization is set for the system so that any boosts (which reduce headroom in the equalizer) are taken into account.
For each device, make sure it is the output and not the input that is clipping.
Always make sure that the any limiters in the system go into limiting prior to anything else in the system going into clipping. In this way the limiter, rather than a clipped signal somewhere else in the signal chain, will control the system’s maximum output.
Remove the test signal, turn off all equipment and reconnect the loudspeaker(s) and the system is ready for level balancing, assuming you have more than one loudspeaker.
If you have only one loudspeaker the system should be ready for use.
Residual Noise
Horn loaded loudspeakers and drivers have high sensitivities which means that they produce a relatively high output for a given electrical input.
This includes the residual electronic noise of an audio system.
By setting gain structure properly and using high quality professional electronics, this noise should be at or near inaudibility.
Among the noisier electronic devices are 16 bit digital devices such as some signal delays.
They have signal to noise ratios that are only 90 dB. However, if the gain structure is set correctly this means that, for example, if the system can produce 120 dB SPL at maximum output, the residual noise should be about 30 dB SPL.
This would be acceptable for a quiet recording studio. If residual noise is a problem, gain structure is usually the culprit – it is never the loudspeakers.
Level Balancing
Once the system gain structure is set, the level balances can be adjusted. This may mean the levels between HF and LF sections of a biamplified loudspeaker, a full-range loudspeaker to subwoofer level, levels between multiple loudspeakers, or between main and delayed loudspeaker array.
The idea is to make the system sound the best it can without using any equalization. This may be done using acoustic test equipment such as an RTA (real-time analyzer), TEF analyzer or similar. You must always determine the final level balance by listening to a variety of known program material.
Level balancing can also be done entirely by ear if acoustic test equipment is not available. In any case, the preferred method of adjusting levels for balancing is to use the amplifier input level controls.
Important:
In order to maintain the system’s dynamic range that was maximized by setting the proper gain structure, do not turn up the input level controls of any of the amplifiers.
For example, if you decide that a subwoofer is not loud enough, do not turn up the input level control of its amplifier. Instead, turn down the input level of the full-range loudspeaker’s amplifier.
Equalization
Once level balances are set, you can then equalize the loudspeaker(s), if necessary. Community loudspeakers are optimized for highly accurate and well balanced reproduction “out-of-the-box”.
Generally, equalization should only be needed to eliminate difficult feedback frequencies or to adapt the system to a difficult acoustic environment.
You should not need more than a few dB of boost or cut equalization for any particular range of frequencies. The best equalization techniques involve cutting rather than boosting frequencies.
Final Results
Once the above procedures are followed, your loudspeaker should reproduce audio cleanly, clearly and with all frequencies in good balance.
Noise should not be audible and you should be able to drive the amplifier(s) in the system to maximum output on normal program material with no significant distortion or other undesirable sound. If limiters are used, the onset of limiting should occur just before any amplifier clipping.
Friday, January 20, 2012
AV: How Do You Set System Gain Structure? - Pro Sound Web
How Do You Set System Gain Structure?
Gain structuring for a system occurs in the signal processing chain between the mixer or another signal source and the power amplifiers.
June 07, 2011, by Chuck McGregor
gain structure
Realistically, audio signals at or near the noise floor of a system are not useful because the signal will not be significantly louder than the noise.
Therefore, some minimum usable level must be assumed below which the electronic noise is considered objectionable.
A signal to noise ratio of 20 dB is considered minimally acceptable for good intelligibility.
For a high quality system 30 dB would be a better figure to use. Using this value, the range from this minimum signal level (30 dB above the noise floor) to the clipping level is the usable signal range window for the system (also called the dynamic range in my way of thinking).
However, for purposes of this paper, the maximum output to noise floor is used as the dynamic range.
Every audio system with more than one electronic component has a “system gain structure”. Gain structuring for a system occurs in the signal processing chain between the mixer or another signal source and the power amplifiers.
One usual scenario is to set all the signal processors to unity gain and turn the amplifier inputs to maximum. Unfortunately as you will see, given the different maximum outputs and noise levels of typical signal processors, this method will may not come close to the best gain structure.
We will be dealing with the signal voltage levels on the interconnecting cables from the output of the mixer (or signal source if there is no mixer) up to the input of the amplifier. For the convenience of using simple numbers, this analysis uses relative dB, as a voltage ratio where dB = 20 x log (V1/V2), and dBu, where 0 dBu = 0.775V. V1 and V2 are simply two voltages.
To set proper gain structure, the interconnections between devices must be constant voltage interfaces. This means an output device’s voltage at any point in time is unaffected by whether or not it is connected to the device(s) it is driving.
This type of interface is characterized by the output impedance of a device being 1/10 or less of its load. For example, if the output impedance is 100 Ohms, the total load it drives must be 1000 Ohms or greater. Virtually all professional audio equipment meets this criterion when a single device drives only one other device.
However when one device drives multiple devices, such as a mixer feeding a number of power amplifiers, this may not be true. In this case a distribution amplifier may be needed to divide the load between its multiple outputs.
The last thing to consider is the power handling of the loudspeaker(s).
As long as the amplifier does not exceed the loudspeaker’s power handling capability and the system is operated without clipping, you should never blow a properly manufactured loudspeaker.
The safest criteria to use in selecting an amplifier is the RMS rating of the loudspeaker.
In reality, most loudspeakers can handle peak signals in excess of this rating.
A reasonable choice is an amplifier whose rating that is 2 times (+3 dB) the RMS rating of the loudspeaker. The RMS sine wave used to rate amplifiers has an inherent peak power component of 3 dB.
So this all works out to a 6 dB allowance for power peaks over the loudspeaker’s RMS rating. This is a pretty safe figure for the way most professional loudspeakers are rated (pink noise with a 6 dB peak factor) and given the peak to RMS content of most audio signals.
However, sustained sine wave signals from the likes of a synthesizer could exceed the loudspeakers RMS capability by 3 dB without clipping the system. If you expect these kinds of signals and you expect to drive the system to maximum output levels with them, use the loudspeaker’s RMS rating as the power rating for the amplifier.
With these basics in mind, we’re ready to examine how to achieve proper gain structure in detail.
Picturing Gain Structure
Before you get out your equipment and start setting gain structure you have to learn just what it is you are trying to accomplish. Go through the following “on paper” analysis of a typical system. After you understand this can you appreciate where to actually set the controls on equipment to achieve optimum gain structure.
Figure 1 shows a simple system consisting of six pieces of equipment. The device clip level (maximum output) is listed for each device as published by the manufacturer. For this example, all devices between the mixer output and the amplifier input are set for unity gain and the amplifier input is set for maximum sensitivity.
Each device is represented by what looks like a bar. Rather than a bar, picture it as a tall, narrow window. The maximum output or clipping point from the specifications for each device defines the top of the window using the absolute dBu scale on the right.
The published noise floor (or signal-to-noise ratio) specification below maximum output determines the height of the window. The relative dB scale on the left is used to determine this height. All usable signals must pass between the top and bottom of the window.
However, remember that your low level signals won’t be near the noise floor. Realistically the minimum usable signal is one that is at least 30 dB above the noise floor.
[Click to enlarge]
Figure 1
Next, a horizontal line is drawn across the top of the lowest window (in this case the amplifier).
This is the system clip level, and for the rest of the analysis this line stays in the same place.
Another line is drawn across the highest bottom window sill (in this case the mixer).
The relative dB scale is used to measure the distance in dB between the 1st and 2nd lines. As you can see, it is only 72 dB for this set of devices and gain structure.
That’s equal to the performance of your average consumer cassette deck—and you thought that professional equipment automatically guaranteed a professional grade audio system. Oh well, live and learn!
Now subtract 30 dB to find the “true” dynamic range (30 dB above the noise floor to the clipping level). The result is 42 dB.
Measurements of the maximum dynamic ranges for acoustic instruments and voice yield maximum figures in excess of 40 dB.
This means our system really doesn’t have enough dynamic range to reproduce them.
The Most Common Approach To Gain Structure
As seen in Figure 1 from the absolute scale on the right, the amplifier input sensitivity limits the maximum signal level in all the other devices to +3 dB. Above +3 dB the amplifier will clip—period. It doesn’t matter how much “headroom” is in the mixer, you can’t use it without distorting the amplifier.
Well, you say, the obvious step is to put a pad (usually the amplifier input attenuator) so the amplifier will clip at about the same point as the next least capable device. In this case it is the notch filter. Using a -12 dB pad, the notch filter and amplifier will both clip at once and the signal level will be 12 dB higher through the other devices at the amplifier’s maximum output.
The chart, as seen in Figure 2, was changed from Figure 1 by moving all the device windows (except the amplifier) down by 12 dB using the relative dB scale on the left. A +15 dB signal (the notch filter clip level) is now attenuated to +3 dB by the amplifier’s input attenuator.
[Click to enlarge]
Figure 2
The noise floor line is redrawn through the highest window sill (in this case still the mixer). Because this ends up 12 dB lower than in Figure 2 relative to the system clip level, we see that the system’s overall window height is now 84 dB. This is a 12 dB improvement - much better.
Note that the absolute device clip levels no longer relate to the absolute dB scale except for the amplifier’s input after its input attenuator. Our usable signal range (from 30 dB above the noise floor) is 54 dB. This means our system is now able to squeak out enough range to reproduce the dynamic range of instrumental and vocal sources.
Unfortunately, the mixer is still the primary noise source by 3 dB over the signal delay. However, according to their published specifications, the mixer should have some 6 dB better noise performance than the signal delay.
It should also be obvious the signal delay is the weakest dynamic range link because it has the shortest window. Therefore, we must conclude that there is more that can be done to optimize the system’s gain structure.
You Can Make It Better
To optimize the system, pads or gain must be added at the input of each device so that its clipping level and the clipping level of the preceding device occur at the same point. Think of the following procedure as a graphic picture of what would happen to the signal on a volt meter as you work your way through the system.
To create the chart shown in Figure 3, the windows are shifted up and down as needed so that all the tops are lined up on the system clip level line. To do this, start with Figure 1 and work from left to right in signal flow fashion. If you move a window up you need gain between it and the next device. If you move a window down you need a pad.
[Click to enlarge]
Figure 3
First, move the mixer window down so its top is even with the graphic EQ window. This movement is measured on the relative dB scale, which in this case is -6 dB. Therefore, you need a 6 dB pad at the input of the graphic EQ.
Next, move BOTH the mixer and graphic EQ windows down together so the graphic EQ window is even with the top of the notch filter window. This also turns out to be -6 dB. Therefore, a 6 dB pad is needed between the EQ and notch filter.
Now, move the mixer, graphic EQ and notch filter windows together so the top of the notch filter window is even with the top of the signal delay window. To do this, move ALL the previous devices up 3 dB. This means you need 3 dB of gain between the notch filter and signal delay.
Repeat the process again by moving the windows of the first four devices together so the top of the signal delay window is even with the limiter window. This distance equals 3 dB. This means 3 dB of gain is needed between the signal delay and limiter.
Lastly, you must lower ALL device windows to line up with the input to the amplifier. They are moved the distance between the top of the limiter and the top of the amplifier. In this example the distance is 18 dB. (In the actual system this would usually be done with the amplifier input attenuator.)
After completing all these steps, the tops of the windows end up on the system’s clip level line as shown in Figure 3 (+3 dB on the absolute dB scale). Looking back from inside the amplifier after its input attenuator, all devices appear as though they are clipping at +3 dB. In reality, they are all clipping at their specified device clip levels. If one device is clipping—everything is clipping.
The Results Are Worth It
Measure the distance between the system’s clip level line and the bottom of the shortest window using the left-hand scale. This result is 90 dB which is 18 dB better than the raw system gain structure in Figure 1. The “true” dynamic range, considering a 30 dB above the noise floor signal as the minimum, is now 60 dB.
Also, the primary noise source is now the signal delay. This is the weak link, which agrees exactly with the published specifications for all the devices. It also should be apparent that more of the usable signal window within each device is being used. What a concept.
In some cases, pads or gain will not have any effect on the overall usable signal range. For example, in Figure 3, the 3 dB of gain at the output of the signal delay could be omitted. The 18 dB pad for the amplifier input would become only 15 dB, and the top of the limiter window would end up 3 dB above the system clip level. The key here is that the bottom of its window is still well below the noisiest device (in this case the signal delay).
You can use this reasoning to save yourself the hassle of making up small pads or small amounts of gain. If you omit one of these along the chain you MUST move all the devices preceding it up or down in your chart by the dB of gain or loss that you are omitting. Otherwise, you will not see the effects of the omission on the noise floor.
Background Noise
Now that you have set up proper system gain structure on paper, it is time to hook-up the system and do the same thing for real. Once completed, audibly evaluate the noise floor heard from the speakers. If all is quiet, pack up and go home. If the noise floor is too high, there are two possibilities:
A. The maximum sound level is higher than necessary, which means you over-designed the maximum capability for the system. If this is the case, turn down the amplifier input attenuator. You will lower the noise, and the maximum output level for the system will be reduced by the amount you decide is over-kill.
B. The maximum sound level you can get out of the system IS necessary, which means your system does not have enough usable signal range. You now have three choices; the first two are compromises.
1) Accept the noise and achieve the maximum sound level you need.
2) Turn the amplifier input down to make the noise acceptable. This will, of course, reduce the maximum output level capability for the system. (Sorry, you can’t have it both ways unless you pick choice 3.)
3) Change the primary noise source in the system to something with lower noise performance.
Doing Your Own Analysis
A similar chart for setting up proper system gain can be created for any system.
Using graph paper, make a vertical absolute dB scale from about +30 dB to -120 dB so you can plot increments for 3 dB or less.
The relative dB scale simply uses the same graph increments for plotting and measuring distances in dB. You could also follow this procedure by using some simple math.
If you don’t trust your addition and subtraction, or would rather work with pictures (they are more dramatic and will quickly show errors in your thinking) cut out rectangular paper bars (windows) like those shown in the figures.
The length of each should equal the distance in dB between the device clip level and its noise floor. Be sure to convert noise figures to noise below maximum output.
Write in the clip level for each device on its window. Using these numbers and the absolute dB scale, position the top of each window on the graph paper in signal flow order from left to right. Move these “paper cut-outs” up and down on the chart as outlined above, by measuring the distances using the relative dB scale. You can very quickly determine the necessary pads and gains—probably faster than with a calculator.
A way to check your work is take the maximum output for the first device and subtract the dB for the all the pads and the gain to that number, including the pad before the amplifier. The result should equal the maximum input sensitivity for the amplifier. This calculation should give math mavens an interesting insight into the gain structure process.
Doing It For Real
To actually adjust a system you need to do exactly what you did on paper except you are now doing it for real.
You start from the console output and find out what you need (gain or loss) to adjust its maximum output signal so that it just drives the next device into clipping. And so on.
You don’t need to know the specifications of the equipment. When you go through the system you’ll find out what those specifications are in terms of maximum output levels. As you should have understood by going through the exercise on paper, the noise floors of the equipment will take care of themselves.
Some device (like the signal delay in the above example) will be the weak link. There is nothing you can do to make this better except to replace it with a device with a better maximum output to noise floor window (better signal to noise ratio specification.
Because of production variations and possibly conservative specifications, you may be able to pick up a few more dB of dynamic range by adjusting the pads or gain values you determined on paper. If things are not reasonably close to your on-paper calculations, you have a problem such as bad wiring or a misadjusted or defective device.
What To Adjust
When you set gain in the system, the attenuation or gain needed between devices can be added externally or by using a device’s input level control, if it has one. DO NOT ADJUST THE OUTPUT LEVEL CONTROL ON ANY DEVICE - this should be left at maximum. This is because it is rarely the last thing in the internal circuitry before the output connector. Unlike some input level controls, it usually does NOT adjust actual gain.
Therefore using it will squash the dynamic range in that device’s output stage and you may end up making things worse, even though the signal level is matched up to the next device. Use an output control only if you KNOW ABSOLUTELY that it is a simple attenuator feeding its output connector. The reason it usually is not is that this topology would cause changes in the output impedance when the control is set for anything other than maximum.
Among other things this could would wreak havoc with - guess what - the gain structure. If the device has a noise floor below other devices when you have set the overall gain structure, you can use output gain.
But reduce it only by 3 dB less than the amount between the device’s noise floor and the device that determines the noise floor of the system. This is because if you bring its gain, and hence its noise floor, up to the worst case device its noise will add to the worst case device and give you 3 dB less dynamic range.
The Tools You Need
To find the clip points in a system, you need to use an oscilloscope and a pink noise test signal. There is really no good substitute for this equipment to set gain structure. Sine wave signals are not recommended as they only show one frequency at a time and you can easily miss something.
The pink noise should be full-range (20 Hz - 20 kHz) and have at least a 6 dB peak to average ratio. If you can find one with a 10 dB peak to average ratio, you will more closely simulate real audio signals.
If you must use sine wave signals, you will have to check each and every EQ boost frequency or range of frequencies very carefully.
When measuring electronic crossovers or other frequency response limiting devices, only a full-range pink noise signal will allow you to see full-range signal energy losses easily. (See sections on crossovers and band limited devices.)
If using sine waves you must set the frequency to the center point of each frequency band of the crossover or the center of the band pass for a band limited device.
For simple systems (e.g. no electronic crossover), there is “poor man’s” method where you use a Piezoelectric tweeter and a 400 Hz sine wave to find clip levels.
Basically, you connect the tweeter directly to the output of each device. When the device hits clipping, the tweeter will emit a very noticeable buzzing sound due to the harmonics in the clipped signal.
For high-powered amplifiers, a resistive pad should be used to avoid burning out the tweeter.
This method is detailed more rigorously by Pat Brown of Syn-Aud-Con. You can find this information here.
Doing It
You start the whole procedure by inputting the pink noise test signal into to mixing console. Set it so that it’s output just clips as seen on the oscilloscope.
Make sure it is the output of the mixing console that is clipping. Determine this by reducing the master fader. The clipping should stop. If it doesn’t, you are clipping something before the output fader.
While you’re at this point, note the reading on the output meter. This is a good indication of what the meter will read when you have reached the system’s maximum output after you set its gain structure.
If you are using sine waves this will NOT be a reliable indication.
Once completed, if the system noise levels are low enough, you may want to increase the setting of the amplifier(s) input level control. This will make the mixer more sensitive for operation.
If you reduce the amplifier input level control, something in the front end of the system will clip first. This means the amplifier will not reach full output. But it WILL reproduce that clipped signal and possibly damage the loudspeakers. Either way—if you choose to increase or reduce the amplifier’s input sensitivity from the optimum gain structure setting—you really don’t gain (pun intended) anything.
There is possible exception to this: by reducing the amplifier’s input level control, the output meters on the console will indicate you have reached the system’s maximum output before the amplifier’s clip.
This is useful so that a less than capable mixing engineer will THINK he’s pushing things to the limit but there will still be something left in the amplifiers. This may help protect the loudspeakers but, bear in mind, it will limit the maximum output of the system to something less than it could be.
Note that to reach a system’s maximum output analog Vu meters on mixing consoles may “peg” before the system clips. If you can afford the reduction in dynamic range, operating the system so the meters don’t peg means you’ll never clip the system. Generally, this means you won’t ever blow the loudspeakers assuming the amplifiers are chosen not to exceed the loudspeaker’s maximum ratings.
More Complex Situations
Up to now we’ve looked at a simple systems. Here is where gain structure gets more complicated. However the ideas are exactly the same. You just have to think about what specific pieces of equipment do and/or about more signal paths.
Devices with Gain/Loss and EQs: Parts 1 - 3 assumed devices in the signal chain have no gain (unity gain devices). However, a device may have gain or loss, or you may want to allow for boosts in an EQ, which may be needed to tune the system.
EQ boosts are like adding overall gain to the device. In such cases, as illustrated in Figure 4, input of the device’s window is shifted down below the output of the device’s window. The distance will be the gain in dB for this device or the desirable dB boost you choose for the EQ.
In this case, it is assumed the boosts will be limited to a maximum of 6 dB. Match the top of the window of the preceding device to this line. On the output side you still use the top of the window to match it to the next device.
[Click to enlarge]
Figure 4
When adjusting gain in an actual system, first set up the system with the EQ set to flat. Then make any EQ adjustments.
If all of your EQ is cut only, you can usually leave everything as is. However, if you add ANY EQ boosts, you will then have to redo the gain structure starting from the input to the EQ by finding the new maximum level it can accept without clipping.
This will of course require attenuation at the input to the EQ input. In some instances a device might introduce a loss in signal level.
The procedure is similar except that the output side of the device’s window is shifted to below the input of the device’s window a distance equal to the loss in dB.
Use this new output point to match the device to the top of the window of the following device. In the example it is assumed the limiter threshold is set so the maximum signal through the limiter is 6 dB below its maximum output.
However, on the input side you still use the top of the device’s window for matching to the preceding device’s window. The relative signal level prior to the limiter is 6 dB higher. As shown in Figure 4, everything, including the noise floor is raised 6 dB. The system dynamic range is still determined by the signal delay, because that is still the smallest “window” in the overall picture.
Because the maximum input of the amplifier (after its input attenuator) is still + 3 dB it has remained in the same position throughout
Multiple Signal Paths, Arrays and Delays
Another variation in this procedure is when a system has several branches, such as a mixer feeding multiple sub-systems. You have to separately analyze each branch and include in each analysis the source common to all branches (the mixer in the example Figure 5).
This will automatically optimize the system so that the common source and all the branches clip at once. To do this, the mixer In Figure 5 must feed each branch through a separate pad. Note that the dynamic range is different in each branch.
[Click to enlarge]
Figure 5
To balance the multiple branch systems acoustically in the actual system, you will probably need different operating levels in the branches than what the optimized electronic gain structure provides.
An example would be a central cluster with delayed balcony speakers. To balance operating levels in these instances, use the branch that is lowest in acoustic level as your reference branch (i.e. the one you are itching to turn up because it isn’t loud enough - but don’t touch that dial). Use the input attenuation on the amplifiers for each of the OTHER branches.
This will reduce their output levels and achieve proper acoustic balance with the reference branch. This will also have the effect of lowering the noise levels and reducing the maximum capability of the other branches. In this case, less capability is acceptable because you have determined that the maximum capability can’t be used in these branches unless you drive the reference branch into clipping.
However, if you find, for example, that you have to significantly reduce the maximum output capability of the central cluster so you won’t clip the balcony system, then your balcony system is under-powered. Instead of attenuating the central cluster, you could add gain prior to the balcony system amplifiers (or “unattenuate” the amplifier input).
While this will balance the system, the balcony amplifiers will be driven into clipping before the central cluster amplifiers.
In this situation, the only way you can have your cake and eat it too, is to increase the size of the balcony amplifier, which translates to more voltage (power) capability for the balcony speakers.
You will not spot this problem by analyzing the electronic gain structure.
This could only have been spotted on paper with proper analysis of the acoustic output for each branch based on loudspeaker sensitivities and listening distances.
Electronic Crossovers
Electronic crossovers require special attention. Consider a full-range signal with equal energy per octave (e.g. pink noise). A crossover will divide the total energy of such a signal among two or more frequency bands. This causes an inherent signal loss at each band-limited output, compared to the full-range crossover input signal.
In effect, crossovers are NOT unity gain devices when fed a full-range signal. You can approximate these losses by calculating how much of the total energy is in each frequency band by using the following procedure:
Example: A 3-way crossover with frequency bands of 50 Hz - 125 Hz, 125 Hz - 500 Hz, 500 Hz -10 kHz.
1) Multiply the lowest frequency in each band by 2 until you get to the highest frequency for that band. The number of times you multiplied = the number of octaves. Round off the results for each band to the nearest whole octave [= 1, 2, 4].
2) Add up the total octaves from all bands [= 7].
3) Divide the octaves in each band by the total octaves [= 0.14, 0.29, 0.57]
4) Push the LOG key for each result [= -0.9, -0.6, -0.2].
5) Multiply each result by 10 to find the approximate losses [= -9 dB, -6 dB, -2 dB].
Note the low frequency output is down almost 10 dB. That is why many systems have problems achieving enough drive levels for the subwoofers.
Now you must draw horizontal lines on the output side of the crossover’s window. Draw these lines at a distance below the top of the window equal to the loss in dB for each output as found above. This line for each crossover output is used to match up the crossover window to the top of the window of the device it feeds (usually an amplifier).
In the example, a different pad would be needed for each output (assuming the amplifiers have equal input sensitivities). The top of the window of the device feeding the crossover is still matched to the top of the crossover’s window.
In the actual system, the amplifier input levels are adjusted to acoustically balance the system similarly to a multiple branch system. Use the frequency band that you want to turn up the most - typically the subwoofer (but of course you won’t turn it up - right?) as the reference output. Balance the other bands to it by turning DOWN their amplifier input level controls.
Once you have the system balanced to your acoustical liking, you may find that amplifier input level controls, in particular for horn amplifies, may be set too low for them to reach full output - even with a single frequency sine wave in their pass band. You can increase all the amplifier input level controls by the same amount to get some or all of this unusable capability back for limited frequency range signals.
Keep in mind, however, that this will have two consequences: It will raise the acoustic noise floor of the system and the capability for full-range signals will remain the same. However, some amplifiers will clip before the signal processing in the system.
This is another situation where you must accept a compromise or change amplifier sizes to get a better match in gain and capability between the different frequency bands.
Other Band-Limited Devices
There is a more general case, similar to the crossover scenario.
If you have full-range signals at the input of a device that limits the frequency response—such as with high or low pass filters—there will be an energy loss from its input to output.
Calculate this loss using the same procedure outlined in the previous section on electronic crossovers.
The significant energy of full range music signals effectively spans about 9 octaves (approximately 30 Hz to 15 kHz).
Example: An under balcony system band limited from 150 Hz to 5 kHz.
1) Multiply the lowest frequency limit of the device by 2 until you get to the highest frequency limit for the device. The number of times you multiplied = the number of octaves. Round off the results to the nearest whole octave [= 5].
2) Divide the number of octaves by the 9 full-range octaves [= 0.56].
3) Push the LOG key for this result [= -0.3].
4) Multiply this result by 10 to find the approximate loss [= -3 dB].
Now you must draw a horizontal line on the output side of the device’s window. The line is drawn at a distance below the top of the window equal to the loss in dB as found in #4 above. This line is used to match up the device’s window to the top of the window of the following device.
System Limiting
The purpose of a system limiter in a properly gain structured system is to prevent any signals from exceeding the system’s maximum level. As such, it is used as an “emergency” device meaning it is intended to provide a hard, never-to-exceed maximum output level.
Limiter/compressors with soft-knee thresholds are not as ideal for protection. You really want something that doesn’t do anything up to a certain point then stops any further increase cold in its tracks. Because you need some margin between the device’s maximum input and its limiting threshold they are a bit tricky to implement properly without compromising the system’s dynamic range.
Just as with any other device you must introduce the limiter using its input, output, noise floor specifications, and gain setting just as with any other device in the system. Because of the way they work, the threshold setting is used as the maximum input.
For proper functioning the threshold should be set to at least 3 dB lower than the maximum output signal from the device preceding it. The limiter’s output gain should be used to adjust its maximum output at threshold to about 2 dB below the input level of the device it feeds. This allows a little “margin for error” in the protection.
If you think about it, the only point you can put a limiter in a properly gain structured system that will truly work perfectly is at the output of the signal source. It would be set so that the input level to system would never allow the first device it feeds to clip. This is technically practical if only one signal source is used at a time (i.e. not mixed with others).
Therefore, if you are simply switching between multiple input sources, put your limiting device at the output of the switcher. The system sees one input source and really doesn’t care which one it is and no input signal can drive the system into clipping.
With multiple mixed sources and a properly set system gain structure, the next best place to put the device is at the output of the mixer.
This is because any mixer output voltage at any frequency exceeding the system’s maximum output will clip the system somewhere.
With multiple branch systems you might think to use a limiter on each branch. But with proper gain structure they would either all work at once or compress some part(s) of the system and not others, thus upsetting the acoustic balance.
Thus a single limiter for each main output that is controlled by the operator makes the most sense.
“Controlled by the operator” means outputs such as a separate sub-woofer output where the acoustic balance is actively “mixed” by the operator based on the input signal content.
You should make sure the operator can see when a threshold is exceeded to avoid clipping the mixer.
If the noise floor of the mixer is low enough compared to the other devices in the system you can allow more than the 3 dB margin the mixer has above the limiter’s threshold. You can do this by reducing a pad between the mixer and the limiter. You can also lower the limiter’s threshold level (and increase the output of the limiter by the same amount) if the noise floor of the limiter allows this.
Because it is used as a “hard-line” device, you should set the compression ratio to maximum (10:1 or higher if available). As to any attack and release settings, they do not affect the gain structure. However, as the limiter is intended to function only as an emergency protection device, there is every reason to use the fastest attack and release times. You are not going for sound quality here; you are protecting the system from any overdrive.
So get into and out of protection as fast as possible. If you think sound quality IS important, then you are not thinking correctly. What you should have thought about is a more powerful system that would rarely be pushed into limiting. In other words if the system is constantly pushed into limiting, it is under-designed.
Summary
Gain structure is only a problem because invariably we use equipment with different input/output capabilities and noise floors. There is no easy way to properly set system gain except to analyze each device in relation to the device that feeds it and in relation to the entire signal path.
Fortunately, the little information you need is readily available on equipment specification sheets. By working out proper gain structure on paper before you purchase and wire up the equipment, you can spot potential problems and make appropriate substitutions.
In any case, with the gain properly structured, you can make significant, or in some cases, spectacular improvements in the system’s dynamic range and its noise floor. In the example system, 18 dB is certainly a spectacular improvement.
Full Article with Graphics:
AV: How Do You Set System Gain Structure? - Pro Sound Web
Gain structuring for a system occurs in the signal processing chain between the mixer or another signal source and the power amplifiers.
June 07, 2011, by Chuck McGregor
gain structure
Realistically, audio signals at or near the noise floor of a system are not useful because the signal will not be significantly louder than the noise.
Therefore, some minimum usable level must be assumed below which the electronic noise is considered objectionable.
A signal to noise ratio of 20 dB is considered minimally acceptable for good intelligibility.
For a high quality system 30 dB would be a better figure to use. Using this value, the range from this minimum signal level (30 dB above the noise floor) to the clipping level is the usable signal range window for the system (also called the dynamic range in my way of thinking).
However, for purposes of this paper, the maximum output to noise floor is used as the dynamic range.
Every audio system with more than one electronic component has a “system gain structure”. Gain structuring for a system occurs in the signal processing chain between the mixer or another signal source and the power amplifiers.
One usual scenario is to set all the signal processors to unity gain and turn the amplifier inputs to maximum. Unfortunately as you will see, given the different maximum outputs and noise levels of typical signal processors, this method will may not come close to the best gain structure.
We will be dealing with the signal voltage levels on the interconnecting cables from the output of the mixer (or signal source if there is no mixer) up to the input of the amplifier. For the convenience of using simple numbers, this analysis uses relative dB, as a voltage ratio where dB = 20 x log (V1/V2), and dBu, where 0 dBu = 0.775V. V1 and V2 are simply two voltages.
To set proper gain structure, the interconnections between devices must be constant voltage interfaces. This means an output device’s voltage at any point in time is unaffected by whether or not it is connected to the device(s) it is driving.
This type of interface is characterized by the output impedance of a device being 1/10 or less of its load. For example, if the output impedance is 100 Ohms, the total load it drives must be 1000 Ohms or greater. Virtually all professional audio equipment meets this criterion when a single device drives only one other device.
However when one device drives multiple devices, such as a mixer feeding a number of power amplifiers, this may not be true. In this case a distribution amplifier may be needed to divide the load between its multiple outputs.
The last thing to consider is the power handling of the loudspeaker(s).
As long as the amplifier does not exceed the loudspeaker’s power handling capability and the system is operated without clipping, you should never blow a properly manufactured loudspeaker.
The safest criteria to use in selecting an amplifier is the RMS rating of the loudspeaker.
In reality, most loudspeakers can handle peak signals in excess of this rating.
A reasonable choice is an amplifier whose rating that is 2 times (+3 dB) the RMS rating of the loudspeaker. The RMS sine wave used to rate amplifiers has an inherent peak power component of 3 dB.
So this all works out to a 6 dB allowance for power peaks over the loudspeaker’s RMS rating. This is a pretty safe figure for the way most professional loudspeakers are rated (pink noise with a 6 dB peak factor) and given the peak to RMS content of most audio signals.
However, sustained sine wave signals from the likes of a synthesizer could exceed the loudspeakers RMS capability by 3 dB without clipping the system. If you expect these kinds of signals and you expect to drive the system to maximum output levels with them, use the loudspeaker’s RMS rating as the power rating for the amplifier.
With these basics in mind, we’re ready to examine how to achieve proper gain structure in detail.
Picturing Gain Structure
Before you get out your equipment and start setting gain structure you have to learn just what it is you are trying to accomplish. Go through the following “on paper” analysis of a typical system. After you understand this can you appreciate where to actually set the controls on equipment to achieve optimum gain structure.
Figure 1 shows a simple system consisting of six pieces of equipment. The device clip level (maximum output) is listed for each device as published by the manufacturer. For this example, all devices between the mixer output and the amplifier input are set for unity gain and the amplifier input is set for maximum sensitivity.
Each device is represented by what looks like a bar. Rather than a bar, picture it as a tall, narrow window. The maximum output or clipping point from the specifications for each device defines the top of the window using the absolute dBu scale on the right.
The published noise floor (or signal-to-noise ratio) specification below maximum output determines the height of the window. The relative dB scale on the left is used to determine this height. All usable signals must pass between the top and bottom of the window.
However, remember that your low level signals won’t be near the noise floor. Realistically the minimum usable signal is one that is at least 30 dB above the noise floor.
[Click to enlarge]
Figure 1
Next, a horizontal line is drawn across the top of the lowest window (in this case the amplifier).
This is the system clip level, and for the rest of the analysis this line stays in the same place.
Another line is drawn across the highest bottom window sill (in this case the mixer).
The relative dB scale is used to measure the distance in dB between the 1st and 2nd lines. As you can see, it is only 72 dB for this set of devices and gain structure.
That’s equal to the performance of your average consumer cassette deck—and you thought that professional equipment automatically guaranteed a professional grade audio system. Oh well, live and learn!
Now subtract 30 dB to find the “true” dynamic range (30 dB above the noise floor to the clipping level). The result is 42 dB.
Measurements of the maximum dynamic ranges for acoustic instruments and voice yield maximum figures in excess of 40 dB.
This means our system really doesn’t have enough dynamic range to reproduce them.
The Most Common Approach To Gain Structure
As seen in Figure 1 from the absolute scale on the right, the amplifier input sensitivity limits the maximum signal level in all the other devices to +3 dB. Above +3 dB the amplifier will clip—period. It doesn’t matter how much “headroom” is in the mixer, you can’t use it without distorting the amplifier.
Well, you say, the obvious step is to put a pad (usually the amplifier input attenuator) so the amplifier will clip at about the same point as the next least capable device. In this case it is the notch filter. Using a -12 dB pad, the notch filter and amplifier will both clip at once and the signal level will be 12 dB higher through the other devices at the amplifier’s maximum output.
The chart, as seen in Figure 2, was changed from Figure 1 by moving all the device windows (except the amplifier) down by 12 dB using the relative dB scale on the left. A +15 dB signal (the notch filter clip level) is now attenuated to +3 dB by the amplifier’s input attenuator.
[Click to enlarge]
Figure 2
The noise floor line is redrawn through the highest window sill (in this case still the mixer). Because this ends up 12 dB lower than in Figure 2 relative to the system clip level, we see that the system’s overall window height is now 84 dB. This is a 12 dB improvement - much better.
Note that the absolute device clip levels no longer relate to the absolute dB scale except for the amplifier’s input after its input attenuator. Our usable signal range (from 30 dB above the noise floor) is 54 dB. This means our system is now able to squeak out enough range to reproduce the dynamic range of instrumental and vocal sources.
Unfortunately, the mixer is still the primary noise source by 3 dB over the signal delay. However, according to their published specifications, the mixer should have some 6 dB better noise performance than the signal delay.
It should also be obvious the signal delay is the weakest dynamic range link because it has the shortest window. Therefore, we must conclude that there is more that can be done to optimize the system’s gain structure.
You Can Make It Better
To optimize the system, pads or gain must be added at the input of each device so that its clipping level and the clipping level of the preceding device occur at the same point. Think of the following procedure as a graphic picture of what would happen to the signal on a volt meter as you work your way through the system.
To create the chart shown in Figure 3, the windows are shifted up and down as needed so that all the tops are lined up on the system clip level line. To do this, start with Figure 1 and work from left to right in signal flow fashion. If you move a window up you need gain between it and the next device. If you move a window down you need a pad.
[Click to enlarge]
Figure 3
First, move the mixer window down so its top is even with the graphic EQ window. This movement is measured on the relative dB scale, which in this case is -6 dB. Therefore, you need a 6 dB pad at the input of the graphic EQ.
Next, move BOTH the mixer and graphic EQ windows down together so the graphic EQ window is even with the top of the notch filter window. This also turns out to be -6 dB. Therefore, a 6 dB pad is needed between the EQ and notch filter.
Now, move the mixer, graphic EQ and notch filter windows together so the top of the notch filter window is even with the top of the signal delay window. To do this, move ALL the previous devices up 3 dB. This means you need 3 dB of gain between the notch filter and signal delay.
Repeat the process again by moving the windows of the first four devices together so the top of the signal delay window is even with the limiter window. This distance equals 3 dB. This means 3 dB of gain is needed between the signal delay and limiter.
Lastly, you must lower ALL device windows to line up with the input to the amplifier. They are moved the distance between the top of the limiter and the top of the amplifier. In this example the distance is 18 dB. (In the actual system this would usually be done with the amplifier input attenuator.)
After completing all these steps, the tops of the windows end up on the system’s clip level line as shown in Figure 3 (+3 dB on the absolute dB scale). Looking back from inside the amplifier after its input attenuator, all devices appear as though they are clipping at +3 dB. In reality, they are all clipping at their specified device clip levels. If one device is clipping—everything is clipping.
The Results Are Worth It
Measure the distance between the system’s clip level line and the bottom of the shortest window using the left-hand scale. This result is 90 dB which is 18 dB better than the raw system gain structure in Figure 1. The “true” dynamic range, considering a 30 dB above the noise floor signal as the minimum, is now 60 dB.
Also, the primary noise source is now the signal delay. This is the weak link, which agrees exactly with the published specifications for all the devices. It also should be apparent that more of the usable signal window within each device is being used. What a concept.
In some cases, pads or gain will not have any effect on the overall usable signal range. For example, in Figure 3, the 3 dB of gain at the output of the signal delay could be omitted. The 18 dB pad for the amplifier input would become only 15 dB, and the top of the limiter window would end up 3 dB above the system clip level. The key here is that the bottom of its window is still well below the noisiest device (in this case the signal delay).
You can use this reasoning to save yourself the hassle of making up small pads or small amounts of gain. If you omit one of these along the chain you MUST move all the devices preceding it up or down in your chart by the dB of gain or loss that you are omitting. Otherwise, you will not see the effects of the omission on the noise floor.
Background Noise
Now that you have set up proper system gain structure on paper, it is time to hook-up the system and do the same thing for real. Once completed, audibly evaluate the noise floor heard from the speakers. If all is quiet, pack up and go home. If the noise floor is too high, there are two possibilities:
A. The maximum sound level is higher than necessary, which means you over-designed the maximum capability for the system. If this is the case, turn down the amplifier input attenuator. You will lower the noise, and the maximum output level for the system will be reduced by the amount you decide is over-kill.
B. The maximum sound level you can get out of the system IS necessary, which means your system does not have enough usable signal range. You now have three choices; the first two are compromises.
1) Accept the noise and achieve the maximum sound level you need.
2) Turn the amplifier input down to make the noise acceptable. This will, of course, reduce the maximum output level capability for the system. (Sorry, you can’t have it both ways unless you pick choice 3.)
3) Change the primary noise source in the system to something with lower noise performance.
Doing Your Own Analysis
A similar chart for setting up proper system gain can be created for any system.
Using graph paper, make a vertical absolute dB scale from about +30 dB to -120 dB so you can plot increments for 3 dB or less.
The relative dB scale simply uses the same graph increments for plotting and measuring distances in dB. You could also follow this procedure by using some simple math.
If you don’t trust your addition and subtraction, or would rather work with pictures (they are more dramatic and will quickly show errors in your thinking) cut out rectangular paper bars (windows) like those shown in the figures.
The length of each should equal the distance in dB between the device clip level and its noise floor. Be sure to convert noise figures to noise below maximum output.
Write in the clip level for each device on its window. Using these numbers and the absolute dB scale, position the top of each window on the graph paper in signal flow order from left to right. Move these “paper cut-outs” up and down on the chart as outlined above, by measuring the distances using the relative dB scale. You can very quickly determine the necessary pads and gains—probably faster than with a calculator.
A way to check your work is take the maximum output for the first device and subtract the dB for the all the pads and the gain to that number, including the pad before the amplifier. The result should equal the maximum input sensitivity for the amplifier. This calculation should give math mavens an interesting insight into the gain structure process.
Doing It For Real
To actually adjust a system you need to do exactly what you did on paper except you are now doing it for real.
You start from the console output and find out what you need (gain or loss) to adjust its maximum output signal so that it just drives the next device into clipping. And so on.
You don’t need to know the specifications of the equipment. When you go through the system you’ll find out what those specifications are in terms of maximum output levels. As you should have understood by going through the exercise on paper, the noise floors of the equipment will take care of themselves.
Some device (like the signal delay in the above example) will be the weak link. There is nothing you can do to make this better except to replace it with a device with a better maximum output to noise floor window (better signal to noise ratio specification.
Because of production variations and possibly conservative specifications, you may be able to pick up a few more dB of dynamic range by adjusting the pads or gain values you determined on paper. If things are not reasonably close to your on-paper calculations, you have a problem such as bad wiring or a misadjusted or defective device.
What To Adjust
When you set gain in the system, the attenuation or gain needed between devices can be added externally or by using a device’s input level control, if it has one. DO NOT ADJUST THE OUTPUT LEVEL CONTROL ON ANY DEVICE - this should be left at maximum. This is because it is rarely the last thing in the internal circuitry before the output connector. Unlike some input level controls, it usually does NOT adjust actual gain.
Therefore using it will squash the dynamic range in that device’s output stage and you may end up making things worse, even though the signal level is matched up to the next device. Use an output control only if you KNOW ABSOLUTELY that it is a simple attenuator feeding its output connector. The reason it usually is not is that this topology would cause changes in the output impedance when the control is set for anything other than maximum.
Among other things this could would wreak havoc with - guess what - the gain structure. If the device has a noise floor below other devices when you have set the overall gain structure, you can use output gain.
But reduce it only by 3 dB less than the amount between the device’s noise floor and the device that determines the noise floor of the system. This is because if you bring its gain, and hence its noise floor, up to the worst case device its noise will add to the worst case device and give you 3 dB less dynamic range.
The Tools You Need
To find the clip points in a system, you need to use an oscilloscope and a pink noise test signal. There is really no good substitute for this equipment to set gain structure. Sine wave signals are not recommended as they only show one frequency at a time and you can easily miss something.
The pink noise should be full-range (20 Hz - 20 kHz) and have at least a 6 dB peak to average ratio. If you can find one with a 10 dB peak to average ratio, you will more closely simulate real audio signals.
If you must use sine wave signals, you will have to check each and every EQ boost frequency or range of frequencies very carefully.
When measuring electronic crossovers or other frequency response limiting devices, only a full-range pink noise signal will allow you to see full-range signal energy losses easily. (See sections on crossovers and band limited devices.)
If using sine waves you must set the frequency to the center point of each frequency band of the crossover or the center of the band pass for a band limited device.
For simple systems (e.g. no electronic crossover), there is “poor man’s” method where you use a Piezoelectric tweeter and a 400 Hz sine wave to find clip levels.
Basically, you connect the tweeter directly to the output of each device. When the device hits clipping, the tweeter will emit a very noticeable buzzing sound due to the harmonics in the clipped signal.
For high-powered amplifiers, a resistive pad should be used to avoid burning out the tweeter.
This method is detailed more rigorously by Pat Brown of Syn-Aud-Con. You can find this information here.
Doing It
You start the whole procedure by inputting the pink noise test signal into to mixing console. Set it so that it’s output just clips as seen on the oscilloscope.
Make sure it is the output of the mixing console that is clipping. Determine this by reducing the master fader. The clipping should stop. If it doesn’t, you are clipping something before the output fader.
While you’re at this point, note the reading on the output meter. This is a good indication of what the meter will read when you have reached the system’s maximum output after you set its gain structure.
If you are using sine waves this will NOT be a reliable indication.
Once completed, if the system noise levels are low enough, you may want to increase the setting of the amplifier(s) input level control. This will make the mixer more sensitive for operation.
If you reduce the amplifier input level control, something in the front end of the system will clip first. This means the amplifier will not reach full output. But it WILL reproduce that clipped signal and possibly damage the loudspeakers. Either way—if you choose to increase or reduce the amplifier’s input sensitivity from the optimum gain structure setting—you really don’t gain (pun intended) anything.
There is possible exception to this: by reducing the amplifier’s input level control, the output meters on the console will indicate you have reached the system’s maximum output before the amplifier’s clip.
This is useful so that a less than capable mixing engineer will THINK he’s pushing things to the limit but there will still be something left in the amplifiers. This may help protect the loudspeakers but, bear in mind, it will limit the maximum output of the system to something less than it could be.
Note that to reach a system’s maximum output analog Vu meters on mixing consoles may “peg” before the system clips. If you can afford the reduction in dynamic range, operating the system so the meters don’t peg means you’ll never clip the system. Generally, this means you won’t ever blow the loudspeakers assuming the amplifiers are chosen not to exceed the loudspeaker’s maximum ratings.
More Complex Situations
Up to now we’ve looked at a simple systems. Here is where gain structure gets more complicated. However the ideas are exactly the same. You just have to think about what specific pieces of equipment do and/or about more signal paths.
Devices with Gain/Loss and EQs: Parts 1 - 3 assumed devices in the signal chain have no gain (unity gain devices). However, a device may have gain or loss, or you may want to allow for boosts in an EQ, which may be needed to tune the system.
EQ boosts are like adding overall gain to the device. In such cases, as illustrated in Figure 4, input of the device’s window is shifted down below the output of the device’s window. The distance will be the gain in dB for this device or the desirable dB boost you choose for the EQ.
In this case, it is assumed the boosts will be limited to a maximum of 6 dB. Match the top of the window of the preceding device to this line. On the output side you still use the top of the window to match it to the next device.
[Click to enlarge]
Figure 4
When adjusting gain in an actual system, first set up the system with the EQ set to flat. Then make any EQ adjustments.
If all of your EQ is cut only, you can usually leave everything as is. However, if you add ANY EQ boosts, you will then have to redo the gain structure starting from the input to the EQ by finding the new maximum level it can accept without clipping.
This will of course require attenuation at the input to the EQ input. In some instances a device might introduce a loss in signal level.
The procedure is similar except that the output side of the device’s window is shifted to below the input of the device’s window a distance equal to the loss in dB.
Use this new output point to match the device to the top of the window of the following device. In the example it is assumed the limiter threshold is set so the maximum signal through the limiter is 6 dB below its maximum output.
However, on the input side you still use the top of the device’s window for matching to the preceding device’s window. The relative signal level prior to the limiter is 6 dB higher. As shown in Figure 4, everything, including the noise floor is raised 6 dB. The system dynamic range is still determined by the signal delay, because that is still the smallest “window” in the overall picture.
Because the maximum input of the amplifier (after its input attenuator) is still + 3 dB it has remained in the same position throughout
Multiple Signal Paths, Arrays and Delays
Another variation in this procedure is when a system has several branches, such as a mixer feeding multiple sub-systems. You have to separately analyze each branch and include in each analysis the source common to all branches (the mixer in the example Figure 5).
This will automatically optimize the system so that the common source and all the branches clip at once. To do this, the mixer In Figure 5 must feed each branch through a separate pad. Note that the dynamic range is different in each branch.
[Click to enlarge]
Figure 5
To balance the multiple branch systems acoustically in the actual system, you will probably need different operating levels in the branches than what the optimized electronic gain structure provides.
An example would be a central cluster with delayed balcony speakers. To balance operating levels in these instances, use the branch that is lowest in acoustic level as your reference branch (i.e. the one you are itching to turn up because it isn’t loud enough - but don’t touch that dial). Use the input attenuation on the amplifiers for each of the OTHER branches.
This will reduce their output levels and achieve proper acoustic balance with the reference branch. This will also have the effect of lowering the noise levels and reducing the maximum capability of the other branches. In this case, less capability is acceptable because you have determined that the maximum capability can’t be used in these branches unless you drive the reference branch into clipping.
However, if you find, for example, that you have to significantly reduce the maximum output capability of the central cluster so you won’t clip the balcony system, then your balcony system is under-powered. Instead of attenuating the central cluster, you could add gain prior to the balcony system amplifiers (or “unattenuate” the amplifier input).
While this will balance the system, the balcony amplifiers will be driven into clipping before the central cluster amplifiers.
In this situation, the only way you can have your cake and eat it too, is to increase the size of the balcony amplifier, which translates to more voltage (power) capability for the balcony speakers.
You will not spot this problem by analyzing the electronic gain structure.
This could only have been spotted on paper with proper analysis of the acoustic output for each branch based on loudspeaker sensitivities and listening distances.
Electronic Crossovers
Electronic crossovers require special attention. Consider a full-range signal with equal energy per octave (e.g. pink noise). A crossover will divide the total energy of such a signal among two or more frequency bands. This causes an inherent signal loss at each band-limited output, compared to the full-range crossover input signal.
In effect, crossovers are NOT unity gain devices when fed a full-range signal. You can approximate these losses by calculating how much of the total energy is in each frequency band by using the following procedure:
Example: A 3-way crossover with frequency bands of 50 Hz - 125 Hz, 125 Hz - 500 Hz, 500 Hz -10 kHz.
1) Multiply the lowest frequency in each band by 2 until you get to the highest frequency for that band. The number of times you multiplied = the number of octaves. Round off the results for each band to the nearest whole octave [= 1, 2, 4].
2) Add up the total octaves from all bands [= 7].
3) Divide the octaves in each band by the total octaves [= 0.14, 0.29, 0.57]
4) Push the LOG key for each result [= -0.9, -0.6, -0.2].
5) Multiply each result by 10 to find the approximate losses [= -9 dB, -6 dB, -2 dB].
Note the low frequency output is down almost 10 dB. That is why many systems have problems achieving enough drive levels for the subwoofers.
Now you must draw horizontal lines on the output side of the crossover’s window. Draw these lines at a distance below the top of the window equal to the loss in dB for each output as found above. This line for each crossover output is used to match up the crossover window to the top of the window of the device it feeds (usually an amplifier).
In the example, a different pad would be needed for each output (assuming the amplifiers have equal input sensitivities). The top of the window of the device feeding the crossover is still matched to the top of the crossover’s window.
In the actual system, the amplifier input levels are adjusted to acoustically balance the system similarly to a multiple branch system. Use the frequency band that you want to turn up the most - typically the subwoofer (but of course you won’t turn it up - right?) as the reference output. Balance the other bands to it by turning DOWN their amplifier input level controls.
Once you have the system balanced to your acoustical liking, you may find that amplifier input level controls, in particular for horn amplifies, may be set too low for them to reach full output - even with a single frequency sine wave in their pass band. You can increase all the amplifier input level controls by the same amount to get some or all of this unusable capability back for limited frequency range signals.
Keep in mind, however, that this will have two consequences: It will raise the acoustic noise floor of the system and the capability for full-range signals will remain the same. However, some amplifiers will clip before the signal processing in the system.
This is another situation where you must accept a compromise or change amplifier sizes to get a better match in gain and capability between the different frequency bands.
Other Band-Limited Devices
There is a more general case, similar to the crossover scenario.
If you have full-range signals at the input of a device that limits the frequency response—such as with high or low pass filters—there will be an energy loss from its input to output.
Calculate this loss using the same procedure outlined in the previous section on electronic crossovers.
The significant energy of full range music signals effectively spans about 9 octaves (approximately 30 Hz to 15 kHz).
Example: An under balcony system band limited from 150 Hz to 5 kHz.
1) Multiply the lowest frequency limit of the device by 2 until you get to the highest frequency limit for the device. The number of times you multiplied = the number of octaves. Round off the results to the nearest whole octave [= 5].
2) Divide the number of octaves by the 9 full-range octaves [= 0.56].
3) Push the LOG key for this result [= -0.3].
4) Multiply this result by 10 to find the approximate loss [= -3 dB].
Now you must draw a horizontal line on the output side of the device’s window. The line is drawn at a distance below the top of the window equal to the loss in dB as found in #4 above. This line is used to match up the device’s window to the top of the window of the following device.
System Limiting
The purpose of a system limiter in a properly gain structured system is to prevent any signals from exceeding the system’s maximum level. As such, it is used as an “emergency” device meaning it is intended to provide a hard, never-to-exceed maximum output level.
Limiter/compressors with soft-knee thresholds are not as ideal for protection. You really want something that doesn’t do anything up to a certain point then stops any further increase cold in its tracks. Because you need some margin between the device’s maximum input and its limiting threshold they are a bit tricky to implement properly without compromising the system’s dynamic range.
Just as with any other device you must introduce the limiter using its input, output, noise floor specifications, and gain setting just as with any other device in the system. Because of the way they work, the threshold setting is used as the maximum input.
For proper functioning the threshold should be set to at least 3 dB lower than the maximum output signal from the device preceding it. The limiter’s output gain should be used to adjust its maximum output at threshold to about 2 dB below the input level of the device it feeds. This allows a little “margin for error” in the protection.
If you think about it, the only point you can put a limiter in a properly gain structured system that will truly work perfectly is at the output of the signal source. It would be set so that the input level to system would never allow the first device it feeds to clip. This is technically practical if only one signal source is used at a time (i.e. not mixed with others).
Therefore, if you are simply switching between multiple input sources, put your limiting device at the output of the switcher. The system sees one input source and really doesn’t care which one it is and no input signal can drive the system into clipping.
With multiple mixed sources and a properly set system gain structure, the next best place to put the device is at the output of the mixer.
This is because any mixer output voltage at any frequency exceeding the system’s maximum output will clip the system somewhere.
With multiple branch systems you might think to use a limiter on each branch. But with proper gain structure they would either all work at once or compress some part(s) of the system and not others, thus upsetting the acoustic balance.
Thus a single limiter for each main output that is controlled by the operator makes the most sense.
“Controlled by the operator” means outputs such as a separate sub-woofer output where the acoustic balance is actively “mixed” by the operator based on the input signal content.
You should make sure the operator can see when a threshold is exceeded to avoid clipping the mixer.
If the noise floor of the mixer is low enough compared to the other devices in the system you can allow more than the 3 dB margin the mixer has above the limiter’s threshold. You can do this by reducing a pad between the mixer and the limiter. You can also lower the limiter’s threshold level (and increase the output of the limiter by the same amount) if the noise floor of the limiter allows this.
Because it is used as a “hard-line” device, you should set the compression ratio to maximum (10:1 or higher if available). As to any attack and release settings, they do not affect the gain structure. However, as the limiter is intended to function only as an emergency protection device, there is every reason to use the fastest attack and release times. You are not going for sound quality here; you are protecting the system from any overdrive.
So get into and out of protection as fast as possible. If you think sound quality IS important, then you are not thinking correctly. What you should have thought about is a more powerful system that would rarely be pushed into limiting. In other words if the system is constantly pushed into limiting, it is under-designed.
Summary
Gain structure is only a problem because invariably we use equipment with different input/output capabilities and noise floors. There is no easy way to properly set system gain except to analyze each device in relation to the device that feeds it and in relation to the entire signal path.
Fortunately, the little information you need is readily available on equipment specification sheets. By working out proper gain structure on paper before you purchase and wire up the equipment, you can spot potential problems and make appropriate substitutions.
In any case, with the gain properly structured, you can make significant, or in some cases, spectacular improvements in the system’s dynamic range and its noise floor. In the example system, 18 dB is certainly a spectacular improvement.
Full Article with Graphics:
AV: How Do You Set System Gain Structure? - Pro Sound Web
Subscribe to:
Posts (Atom)