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Old 29th March 2011, 10:57 AM  
Pano is offline Pano  United States
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Location: Milliways
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Default What is Gain Structure?

Gain structure (AKA Gain Staging) is a concept that gets talked about a lot in pro audio, but most home audio folks have never heard of it. Understanding gain structure can help you get the cleanest signal possible out of your system and avoid some nasty things. Things like noise and clipping,...

Last edited by Variac; 1st April 2011 at 11:34 PM.
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16th April 2011
Conrad Hoffman
diyAudio Member
Pano, it may well be. It seems that audio people, pro rf people, amateur hams, general ee types, electricians and physicists all have their own special vocabulary to describe exactly the same things! Interestingly, their fundamental understandings of how things work are also often different.
May the root sum of the squares of the Forces be with you.
16th April 2011
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AUDIODH
diyAudio Member
I've been in audio for over thirty years..as well as electronics. I have always heard and used gain staging...never the others mentioned. Perhaps it's the EE schooling, or maybe it's geographical...but I understand either term, whether or not I agree what is right or wrong -I only have one each.
16th April 2011
t3t4
diyAudio Member
Absolutely yes!

Quote:
Yes, a main component of gain structure is that we almost always have too much gain. It's there "just in case" but too often leads to trouble. We often just don't need much.

As Mr. Hiraga was one of my teachers, you could say I come from the Low Gain school of audio. And working in pro audio for years taught me that getting it right is important.
Gain can be/will be the death of us all... Less is better on all counts, but use the power of the amp, not the pre-stage input!

It's a tough game to play, I know. But the more you add component wise, the more distortion you "will" receive. The same is true of gain! Less is more in most if not all cases...

Until we deal with diamond based structures and/or components, we will always have loss and/or distortion. But also until then, keep your gains low with as few a component as possible!

My $0.50 worth.

t3t4
16th April 2011
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danielwritesbac
diyAudio Member
sandyK
Moosefet from Classic Valve Design is a dandy inexpensive little project with a small gain and some triode like harmonic filtering ability to increase intelligibility of digital TV, computer audio and the like. I think you can use it only for sources that need a boost so that you don't have to run high gain on everything.

Some people use a compressor limiter device on the TV to bring down the loud commercials and bring up the whispering voices.

I use the Moosefet on the TV because that TV already has AGC to defeat loud commercials. And I use 6n3p current buffer to filter Sony's HD radio just so that radio doesn't sound so awful and clip itself.

So, I'm saying that its okay to have specific preamps for specific sources. Its also inexpensive and its fun. You can sort of tune in each one to your own personal preferences.
16th April 2011
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fotios
diyAudio Member
That is a good article, very well focused on the issue. I have to add only a comment:
Gain is a restriction factor of Bandwidth according to my experience. Stages with unity gain have excellent bandwidth, instead when the gain of a stage is going increasing its bandwidth restricted proportionally.
Best Regards FOTIS ANAGNOSTOU
16th April 2011
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Pano
diyAudio Moderator
Interesting. A quick Google shows both terms being use for the same thing.
May need to edit the article to include the "staging" term, if it's in common use.
16th April 2011
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fotios
diyAudio Member
Another one issue of big importance, is the volume control element (of any type: carbon track pot, opto-resistor, IC etc). When it works like a voltage divider at the input of a stage, it causes non-linearities in high frequencies. When is placed at full (i.e. 0dB thus no signal attenuation) it works like a simple resistor establishing thus a "constant" input impedance combined with the Zin of the following stage, where does not affects the high frequencies. In the past (working for more than two decades with pro audio equipments) i had not noticed this phenomenon, because pro audio signal processing devices are implemented by 99% with ICs which have limited bandwidth. Working the last two years with discrete implemented stages, I have to admit that the volume control part IS A REAL NIGHTMARE to me. Discrete stages with JFETs at input, seems to be not infected from the voltage divider established by a volume control. Instead those with BJTs at input are easily offended by the volume control. It seems to be an issue of the Zin of the stage that follows the volume control, because JFETs present very high Zin compared to BJTs (i am working exclusively with BJTs to the present). I did a lot of experiments on this issue, using always the same test setup: A 10KHz square wave as stimulus and a DSO connected at the output of the stage. Up to now, i have ascertained that a small resistance of the volume pot (up to 10KΩ as much, 1KΩ is "a must") can resolve the problem of high frequencies infection (overshoot or excess rounding of square wave rising edge).
Best Regards FOTIS ANAGNOSTOU
16th April 2011
blondamps
diyAudio Member
I love all you guys! As a fiddler who will NEVER stop trying out different things to keep "scratching the itch" I'm so happy that so many clever, resourceful, motivated people can STILL be doing what so many of us have been doing for 40 years or more and still want to keep doing it. As an oldie who works with youngsters whose only comprehensive understanding stretches to 'digital' audio, to see something that is so clear and commonsense about starting at the input and getting the right result at the output is refreshing indeed. Young hero musicians, PLEASE take note, and many MANY thanks, Panomaniac!
16th April 2011
chaimovitch
diyAudio Member
Hi, I like this thread. In fact I think that I've got some problem with my "gain structure".
One question I'd like to raise is: does one need different gain structures to playback rock/pop/compressed CDs versus acoustical/classical/non compressed CDs?
To put things into practice, using Adobe Audition, a rock track (My Wife/ Who's Next/The Who) has a Peak amplitude of 0dB and an Average RMS amplitude of -11.5dB. A classical song (own recording of a Mozart's Berceuse) has a Peak amplitude of 0dB (the last chord) and an Average RMS amplitude of -42dB during 99% of the track. This makes a difference of 30dB on average perceived loudness with both the tracks peaking at 0dB.
As it's uncomfortable/dangerous to one's ear system to listen to the Who at more than 95dB for long, it is absolutely safe to listen to peaks of a few milliseconds during a acoustical track. Does it mean that one should have a system with a dedicated gain structure to play back compressed music and a different system with a dedicated gain structure to play back acoustical (not compressed) music?
16th April 2011
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Pano
diyAudio Moderator
Darn good question, Chaim. My first thought would be "No" but I'm open to criticism on that point. That Who track is super hot, it's in the category of over compressed mastering that is so disliked today. It's so loud all the time, how would you ever hear any noise? As long as it's not clipping anywhere, you likely won't hear any problems.

The Mozart on the other hand is rather low. -22dB is where I see a lot of classical CDs mastered. But a lullaby should be quiet, after all. That's the one that going to be a challenge. You want to keep that -42dB out of the dirt, but not clip the system or blow out your ears on the final chord. It can be done, but needs some care. The average music level is 133x lower than the peak. Meaning its around 0.015 volts coming out your CD player.

One way to figure out the levels along the path is to work backwards. Let's take an example with the Berceuse. How loud is the peak? Once you have it set were you want it, you could measure a 0dB sine wave at the speaker. Since 0dB is as loud as you're going, that's your reference. And if your CD player is typical, 0dBFS (full scale) is going to be 2 volts RMS or 2.8V peak. Working back from the speaker, to the amp inputs, to the preamp, to the CD player will give you an idea of how much gain (or attenuation) each section has. You'll probably find that you've actually dropped the signal somewhere along the path.

What can be done to optimize it? Without changing any of the components, you might find that you can get a stronger signal out of the preamp and attenuate more at the power amp. Is the power amp far below clipping at the max SPL you want? If so, maybe a lower gain or lower power amp would serve you better.

The first step is to measure and know what the signals are along the path. That will help you get a handle on how well balanced the gain is at each stage.

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