What is Gain Structure?

My rule of thumb is to keep the signal about 6dB below clipping. That can often be impossible or impractical, but it's a goal to shoot for. You don't want to go above that level - unless you're looking to add distortion. Dropping too far below that means you'll pick up noise.

That's it in a nutshell.
Pano,
well said!
The 6 dB headroom is a very good margin to shoot for. If your a live player and/or sound man....that is a big part of your job.
In home theater/stereo systems, that is taken care of in the mastering process. Often times to the point of destroying dynamic range....can you say 6dB overall headroom!
Pano, I see we completely agree on this subject and I hope it helps all the DIY members when gain staging their system.
 
Going back to the first Gain diagram.
If the X4 gain stage were switchable to bypass (X1) or to +6dB (X2) then the system as it was could be adjusted to suit virtually all sources

Hi Andrew,
that is a true statement.
In an ideal world for us tweaks, a continuously variable gain pot is optimum, preferably with an signal/clip indicator for optimum level in the associated device. However, in many systems, that just means more $ to the end user. There is a good argument for a matched system where that engineering has been done up front and isn't a concern.
I sometimes run through as many as ten devices in the signal chain through both analog and digital paths, I have to relay on indicators and my ears to optimize the gain staging.
-Dave
 
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Talking about the gain strcture and the levels accross the signal chain...I miss those days where almost any device had its own VU meter, so you knew where is your signal. Reel-to-reels and cassette decks all had nice analog or digital meters, to keep the signal as far from the noise, yet safe from overload...even good pre-amps had the signal meters, not to mention power amps. Its a pitty that those nice fluorescent VU meters dissapeard, they were so much fun to watch, especially the fast ones. If you see complete systems these days, there are barely any VU meters!
 
Adason,
yes I use to have a visual on the old McIntosh system with those beautiful blue meters and lights while listening to some Frazier Texas Bull Horn loudspeakers. That system rocked! It was in 1970 when I was young enough to truly sit and listen to really good music!
Your system looks nice. I could see you had Bryston power, and from the looks of it, Bryston preamps too! They design great stuff and the whole gain stage blog is a nonissue with quality matched gear as that you have! I'm sure you don't experience noise, clipping, etc with that system.
-Dave
 
It would be interesting indeed to see some other examples of amplifiers that have gain strictly in proportion to output potential. The majority of examples like that may turn out to be high power models (prosound scale).

What are some of the more modest (home scale) exceptions that actually work well?

Are there more good examples of high current output buffers (purpose made for driving speakers) to compare and read up on for reference?
 
Michael
What you are saying is highly desireable.
However, until STBs and DTVs etc. use a gain stage to make up for the <-20dB level used with DTV broadcasts,
( and the level of the L and R from 5.1 Surround broadcasts is even lower)
then many of us are stuck with the need for additional gain, even though it will degrade other sources.
The DTV level via SPDIF is also low, and you really need to take extreme measures to avoid further S/N degradation,
otherwise low level ambience information is severely degraded.
Alex
 
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. :mischiev:
 
Absolutely yes!

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
 
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.
 
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.
 
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).
 
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!
 
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?
 
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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.