What is Gain Structure? - Page 6

29th March 2011, 10:57 AM
Pano diyAudio Moderator Join Date: Oct 2004 Location: SW Florida	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,... Click here to read the entire article Last edited by Variac; 1st April 2011 at 11:34 PM.

19th April 2011

Pano

diyAudio Moderator

LOL. Thinking can hurt sometimes.

Quote:

I have no intention of listening to music on 10 different audio systems, pending on the volume i desire.

That would be BAD gain structure. There is no need to do that at all, it's wildly impractical.
All you need to do is find or design a system that goes as loud as you want, and then a bit more (headroom), then optimize the gain throughout. The volume control is your friend. You just don't want to be stuck at the bottom of its range all the time, or overdrive some of the components while under-driving some others. It's about balance.

19th April 2011

paba

diyAudio Member

good article.

I've often wondered why some "source selector with volume attenator" units have a fixed gain before the volume knob and others fixed gain after the volume knob... I figured it was more for impendance matching either with the previous stage or the next stage.

driving a car keeping your foot hard on the gas and hitting breaks from time to time or driving a car with the foot on the breaks and hitting the gas to move ... very different passenger experience

my signal path:
cd/tuner -> passive transformer attenuator -> SET(8W) -> speakers

21st April 2011

RCruz

nhinstruments

diyAudio Member

Splendid text !

I need a lower gain setup

22nd April 2011

Blues

diyAudio Member

Nicely done article, Michael... a good read for all diyers.

In a domestic system setting, I think we can do away with the gain contributed by preamps. With the current crop of digital Sources and DACs at 2Vrms or more voltage output and not too high an output impedance the active preamp is dispensable. This combination of voltage and impedance specs will have enough current to drive amps to healthy levels of sound even with a lowish voltage gain of 4x (12dB) and loudspeakers in the 88dB/2.83V sensitivity range. Unless of course your domestic setting is a stadium-like great room! And that of course your amps are current capable with your fancied speakers' impedance specs.

Also on headphones, I find running some numbers, that headphone amps too are a waste of gain. Again digital Sources with the standard voltage output with a lowish output impedance coupled to a capable volume control with enough range of levels will be more than enough to go very loud. With Noise Induced Hearing Loss pegged at 85dB and headphones with sensitivities at 98dB/1mW or more, I say we don't need additional gain. I will be posting a spreadsheet soon to prove my point.

All the best,
blues

22nd April 2011

Pano

diyAudio Moderator

Thanks Blues.

Yes, agreed. In most home systems an active preamp isn't needed. Thus the popularity of the integrated amp. Also agree on the headphone amps. The 32 ohm types may need a little voltage gain.

I can say, tho, that I've heard some preamps or line stage with minimal voltage gain but good current gain improve the sound. Using a somewhat low value pot also seems to work well, such as 10K - if your source will allow it.

The real trouble comes when you start getting more devices in line. Things line phono preamps, active crossovers, EQs and such.

24th May 2011

pski

diyAudio Member

I've moved to wifi preamps in music playback the most part. I'm aware of "passive attenuators" vs "preamps" (sometimes in the same box.)

I'm feeding a sub-through/direct to amp connection as well as direct to amplifier (etc.)

I suppose it's more accurate to set the digital player to 100% and use passive attenuation before the amp, but how much am I really suffering, quality wise, using the "digital attenuation" option?

Paul

7th June 2011

wintermute

just another

diyAudio Moderator

Thanks Michael, that makes a lot of sense! Also makes me pleased I'm going down the route of using buffers in my active crossover/preamp (ie there will be no gain).

It does make me want to lower the gain on my chipamp (currently 32X) I can drive it into clipping with only 50% output on my pc sound card. The only problem there is that it is p2p and I didn't make it easy to subsequently modify! In fact I just checked the build pics and I would say pretty close to impossible

Looks like I'm going to have to start on that Baksa!

Tony.

Any intelligence I may appear to have is purely artificial!
Some of my photos

9th June 2011

krivium

diyAudio Member

Pano this is a great article, well and clearly explained.

Pski using digital attenuation lead to different plague...

You are dealing with quantification problems.

If your source file is 16 bit (CD red book standard) you have approximately 65000 steps to express dynamic content of music (approx 90db usable dynamic range). If you attenuate in digital domain you'll loose many of this steps during the process, and digital being referenced to maximum out level (0dbfs=0db full scale) you are loosing definition with quiet sounds and increase quantification noise (digital background noise). With highly dynamic music content with low subjective level (classical music) you are loosing many information along the process. Mainly 'delicate' contents are lost (infos about the recording space, harmonics, ...).

If you are using 24bit source files the problem is less effective: you have approximately 2 000 000 steps allowed for dynamic reproduction ( approx 120db dynamic usable) range. And reference level, 0dbfs, being the same max output level the increase in resolution is focused on the other 'end' of the digital scale, where the 'delicate' informations are. Attenuation in digital domain will reduce again the number of steps allowed for reproduction of dynamic but this will leave many steps for low level information and with increased usable dynamic range, quantification noise level is lower too. So definition is better but still not optimal.

Depending of music style it can be really a problem (low subjective level -rms- as classical music) or something you won't notice too much (high subjective level -highly compressed/high rms- like metal/ hip hop/pop music).

So for me, buffered analog attenuators are better choice quality wise in the gain structure.

9th June 2011

Pano

diyAudio Moderator

Thanks Guys, glad you liked the article.
Tony- yes we often have more gain than we need. My horns are an extreme example as running loud they are 4-6dB below unity gain. All I really need is current gain. Most systems just don't need 30X gain from line level.

krivium: I do think that analog attenuation has its advantages, but I use digital. My player does all manipulation in 64 bit floating, then recalculates that to 24 bits for my sound card. Not a big loss - and I usually don't run more than about 10dB of attenuation anyway. I can not hear a difference between that and good analog attenuation. YMMV.

Take the Speaker Voltage Test!
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23rd June 2011

revboden

diyAudio Member

I start with the average spl required for the program material, calculate the input voltage needed for the amp/s to work in the 10-60% output range. Then.. get to that voltage as early in the signal path as possible, then set the rest of the gains to passthrough. Then, I spend time listening to the background noise with headphones while onsite equipment is turned on and off and adjust my power and cable runs to minimise the interference.