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Gain Structure for Home Theater Discussion Thread

13480 Views 85 Replies 22 Participants Last post by  Maxino1969
Please use this thread for any comments or discussion about my article Gain Structure for Home Theater.

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re: Gain Structure for Home Theater: Getting the Most from Pro Audio Equipment in Your System

I apologize for the length of this, but I figure if you’re going to scorch sacred cows you’d better have the documentation to back it up.
I missed the documentation. Where are the distortion, noise floor, and headroom graphs, and information to back this method up?
re: Gain Structure for Home Theater: Getting the Most from Pro Audio Equipment in Your System

I edited them out because I felt the questions would turn things into a line by line discussion, and I have no desire to do that. I have been down this road many times, and this does not seem like the proper place. That was me, stepping out quietly.

Relevant graphs were presented in Part Three. The “sacred cow” I was referring to was the popular maxed-out signals canard. I’ll go back and change the text to make better sure the point is made.
I still do not follow then. You say that which is true as structures gain has nothing to do with maxed-out signals, and later add "
Gain structure is merely an exercise to insure that the pro amps are getting enough signal to drive them to maximum output.
" this is not so accurate. The idea is to pass the cleanest signal possible along the signal chain. Live sound extends this to the amps, but that is to protect fidelity, and equipment as well. It is not for the sake of getting maximum output.

OdBFS is also the maximum a DVD player will output, since a digital output it’s passing the signal straight along to the AVR. I assume the same can be said about things like cable TV and sat receivers as well.
0dBFS on a disk has a fickle relationship with what you end up with out of the AVR analogue outputs. +0dBFS signals are very common, and are produced by the DAC conversion, bass management, and the master volume control alone, and as a group. There are also oddities with some devices and 0dBFS signals as well where 0dBFS may not be as loud as signals just a few dB lower in intensity. Most every commercial CD released in the last half a dozen years can produce +0dBFS. It's even worse with most mp3s. No one is ripping DVDs in raw format, but I see no reason for it not to be the similar.

And that’s perfectly fine. This is not pro audio. This is home theater. The problem people have been having is thinking the “pro audio mindset” somehow became relevant to them when they added a piece of professional equipment to their home theater system. As thoroughly discussed in Part 2, a pro audio-styled gain structure protocol does not necessarily cross-reference to home audio.
If you put a piece of differently referenced gear into your signal chain, it is now relevant. Structure, protocol, terms, have nothing to do with it. It is basic electronics. Home gear runs on the exact same base principles pro gear does. Once you add a device that does not conform to the same default "home" reference, it pays to think about the things home audio takes for granted as a given.

The “set speakers to max” thing is only for the purposes of setting the amp gains. After that you’re supposed to adjust the relative speaker levels as it’s normally accomplished, with the AVR’s rotating pink noise sound, a calibration disc, etc. Headroom is restored when you turn everything back down and use the system as normal. You weren’t running your system wide open before gain-structuring, and you won’t be afterwards.

But it’s imperative to have the maximum signal on tap when setting the amp gains. Otherwise you will end up with the gains being set higher than they need to be, which can lead to more noise from the amp than you’d get with lower gain settings.

Besides, my “max level” procedure for setting amp gains is virtually identical to what Rane outlines in their gain structure article, which I linked in Part 2. Have you read it?
Starting at the end... If you look at part 4 in the Setting Power Amplifiers section, you see that you need to identify the max clean signal from everything up until that point first. SOP. Just turning up the levels to max isn't the way to go. Most AVR outputs will be distorting and clipping before the max anyway, so it's kind of arbitrary. You will clip the amp before the input sensitivity voltage is reached. You are tying to get around clipping the amp by raising the AVR to levels it should never normally encounter, but doesn't this still leave the amp "with the gains being set higher than they need to be?" This leaves the amount of gain the AVR can add to a channel out of the mix. With 12dB you should be alright, but what about models with only 6dB?

What happens when the same AVR and amp are used in two setups with one having low sensitivity speakers, and the other high? What happens when you switch out an amp of higher/lower power? Now add a Pro EQ, and run though again? Which one was the optimal setup?

It seems like setting up from the end of the signal chain to the front.
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re: Gain Structure for Home Theater: Getting the Most from Pro Audio Equipment in Your System

Let me get a little long winded here. :D

The point you seem to be trying to make is that most AVRs should be able to run a proamp to maximum power with no problems. This is true for most AVRs, especially for AVRs in the middle up range. The problem over the years has been one group that thinks there is never a situation where an AVR can't drive a pro amp properly, another group that has decided no AVR can drive a pro amp properly, and the group of newbies who can't tell who to believe. Do they trust one of these two, or the other guy saying try it out, if it doesn't work add a converter.

The issues with Pro signal processing gear added in is another nut altogether.

The discussion on the home audio forums, as least what I’ve seen in the past ten years, has been that pro gear requires a +4 dBU signal, and if you can’t get that from your home theater pre amp/AVR you’re going to have noise, reduced dynamic range, etc.
Unfortunately, it is true. That doesn't mean it can't work fine that way, but to say it is anything but true is just wrong. Of course you have to understand what +4 dBu is.

Most Pro gear is set up to operate in it's optimal range at 1.228volts balanced (+4 dBu or 0 Vu.) This is not the ceiling though. It is not like 0dBFS on a DVD. This is the nominal signal. It would be like -20dB in the film world (-30dB is the HT equivalent.)

Now lets say the balanced device has an input ceiling of +22dBu (very common,) and your unbalanced out can only do 5v max clean (a common LFE out number.) +22dBu is 10v, so you are 6dB closer to the noise floor than you should be, and have lost 1bit of resolution. If this was a signal processor, and you connected it to a consumer unbalanced amp you just lost another 6dB do to the conversion.

This brings up another thing that is mis-stated all the time. You automatically lose 6dB of headroom and gain going from balanced to unbalanced (unless you have a special cross-coupled output, then you just lose headroom and not gain :dumbcrazy:) not the other way around.

You’re “mixing and matching” the digital dBFS scale with the analog Vrms scale. 0 dBFS is the highest possible digital signal; there is no such thing as “+0dBFS.” If a component somehow adds some boost to the signal in the analog domain (i.e. after the digital-to-analog conversion), that’s of no relevance. The measurable-voltage signal at the AVR’s main pre outputs will reflect that, and any voltage measurement will still be valid.
I was just trying to show that 0dBFS is not always something that scales properly with the rest of the signal. The voltage jump from -10dBFS to 0dBFS could be 10dB, 16dB, or 5dB because of the 0dBFS+ (I was improperly using +0dBFS before) conditions present in all digital to analogue systems. 0dBFS is absolute on the disk, but ones it leaves all bets are off. This is part of the reason why digital test signals are sent a -20dBFS to -10dBFS.

For a clean reference signal (assuming a fully digital connection up to the AVR,) a -6dBFS peak sine should avoid all the Nyquist headache for a single channel test. You would just bump the master volume +6dB to compensate, as this is done after the digital section. For an LFE test you would want an identical -6dBFS wave on the LFE+L+R channels, and use bass management to send everything to the sub out. This will just fill the 5dB digital headroom in the DD processor, and allow a full strength signal out of the preout with the master volume at +6.

This is without getting into distortion levels. Such as, my AVR at 0mvl(corrected) will begin to audibly distort with a simple LFE channel signal of 0dBFS(corrected) without redirected bass with a channel level gain of +4dB of +12dB possible. You can test this with software that has loopback capability, or adjust the power amps sensitivity so that it is only 80dB in-room instead of 120 something. I also found that around +8dB I started to get premature clipping from my amp. Unfortunately, you can not register distortion/clipping with a DMM. It requires measurement.

There is a long and established history on the home audio forums that trying to apply a pro-audio-styled gain structure protocol has caused a lot of confusion, if not out-and-out problems. As an example I recall at least one thread at AVS I came across while researching this, of a guy who had added a Behringer DCX2496 to his system. Following the pro audio protocol, he’d managed to get his levels hot enough to get a good reading on the DCX input meters (forget how he accomplished that). The result: An added 6 dB of noise (by his account), and problems clipping the inputs of his home audio amplifiers. Then there was the case I linked at the end of the article.
That is not how a professional would do it though. :bigsmile: There should be zero clipping. Clipping takes precedence over DNR. He bumped up his levels to get the most DNR out of the signal processor, only to overdrive his -10 dBV amp inputs. He should have accepted the DNR loss through the DCX, or added another line converter after it to reverse the one before.

I expect people have the good sense to know if they are getting distortion and can easily tweak things to get the desired results.
Those people wouldn't need to read this to know how to set things up. ;) The point of structured gain is to never get to the distortion/clipping level, but also maximize DNR of the total system. You can miss the distortion/clipping level, and still have lowered your DNR.
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Re: Gain Structure for Home Theater: Getting the Most from Pro Audio Equipment in Your System

I thought I had posted to this awhile ago, but I guess not? :scratch:

Your posts are very helpful, so don't consider my comments to be purely critical. I simply want to make it more likely that the general reader gets the point.
I didn't find any problems with them. You are absolutely correct about the 1bit in analog comparison. I use it help help explain the difference, as I find more people get the idea better that way, but normally I add a line about it not being technically correct.

I tend to write my posts as a stream of consciousness off the top of my head, in a spare minute or two here and there, and sometimes things get left out. I screw up too, and have no problem with someone pointing it out. ;)

Basically you're re-stating the max-levels theory. However, pro gear, just like consumer, can be operated at any signal level, and a high or low level does not change the noise floor of the component in question. The component’s noise floor is fixed. It’s not going to increase, even if you reduce the signal down to the level of background music, or even to zero.
I'm not talking about max-levels theory here, but it is similar. I was talking about a digital processor there, but lets touch the analog first.

You are correct the noise floor of each individual component does not change. The input to output ratio (or gain) does though. The idea is to have the input as high as possible to get a good SNR through the component. If the signal input is too low you lower this ratio on the output, and you can't get it back. This is the cause of the dreaded hiss. The idea is to have enough signal to avoid this noise creep.

Max-Levelers take this to the extreme, and hook everything up to an oscilloscope, and wring every last bit out of a piece of gear. As a result the manufactures keep adding more headroom, to appeal to these guys. This is fine, but once you signal is high enough to avoid audible hiss it isn't worth much to anyone else.

EDIT:Additional testing confirmed that distortion from the pre amp outputs is a valid issue. Of primary concern, when the amplifier gains are set with using the maximum pre-amp output, the amplifier gains end up being set too low. This means the pre amp has to be turned up further to get the amps going, which most likely will lead to distortion from the pre amp during normal system operations. Appropriate additions have been made to Parts 7 and 9 taking distortion into account, and how to avoid it.
This is a very good step in the right direction. I would also suggest doing the same thing to the sub channel. Depending on the AVR, things can be a bit different between the LFE out, and the other channels.

I shout out to REW would be nice as well, as it can run the test on all your components individually in loopback mode should one get the itch.

Wayne A. Pflughaupt said:
It will work just fine with a consumer signal level with no noise penalty. Any background noise it may have is fixed and will not change with signal levels, so choose your accessories carefully.
This is worded a little oddly. The component noise floor is static, but the component SNR and the total system noise floor is not. If you have a low signal voltage going into a digital processor (anything really) the SNR will be lower when it hits the output stage, than a larger input signal. As a result in the next component in the chain will receive a signal with a lower SNR, and in the end usable DNR.

Lets say you are using a signal with an average level of -20dB. You run it through a component using a low voltage level, and your -20dB is now only 30dB above the noise floor. When you amplify the signal in the next component your -20dB signal is at say 80dB SPL, and the hiss from the processor is now amplified to 50dB SPL.

If you go back and raise the input voltage level 10dB, your -20dB average signal is now leaving the the component at 40dB above the noise floor. When amplified you end up with the same 80dB SPL average level with a 40dB SPL noise floor. This is why you start from the beginning, and work down the chain. You can't make things any better down the chain, only hold it steady, or make it worse.

You don't need to have everything at the input limits, but you do want them to be high enough to keep the noise floor from creeping up on you. This point will be different for every setup, as every piece of gear (including the speakers,) will have a different effect on the overall system response.
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Re: Gain Structure for Home Theater: Getting the Most from Pro Audio Equipment in Your System

I unplugged all my speakers and raised the L and R speaker evels on the AVR to max. I connected a DVD player with a digital coax connection and bipassed all tone and signal processing with the "pure direct" feature. I played the 60hz sine wave track. I turned up my Yamaha and got to about -7, then it promptly shut off. The voltage reading was always about 2-2.4 volts when this happened.

First off, should I be concerned about damaging the receiver? Why is it going into protection mode? Considering max volume is +16.5, I'm surprised that -7 volume is overloading it. I understand that I've also maxed out my L and R levels from 4.5 to 10, so that contributes. But I was really not expecting this shut down to happen. This receiver has given me no trouble before.
This is normal. It is a little low for it to be happening though. You have the AVR trying to put out too much voltage. I would suggest resetting all the channel levels in the AVR to 0, then recalibrate manually with an SPL meter adjusting the amps themselves instead of the AVR. You should be able to run at ~ +4dB with a 0dbFS sinewave for ~2minutes before the cut out. Then once working again, you can try to tweak things out if you want.

As for the numbered section:
1) I wouldn't push my luck.

2) You had the AVR trying to put out more juice that it could handle. See above.

3) Do not try to run sinewaves with the channel levels at max with a high MV on a Yami. It will just shut itself off. This is not a knock against Yami, as I run one myself, but it is an issue you will find with extreme level sinewave testing.

4) Just turn the AVR channel levels back down, and then go from there.

5) I don't think you need to buy anything else. You just need a good tweak on what you have.
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Re: Gain Structure for Home Theater: Getting the Most from Pro Audio Equipment in Your System

Nah, you seem to be mixing what I said in with the digital signal limit issue you seem have a bone about, and that is not what I was talking about. The input meter readout level is irrelevant here.

What I posted is fact, and can be easily demonstrated. The input and out put levels at every stop along the chain will effect the total/final/in room system noise floor. I also stated throughout these posts that each system is different, and each stop along the chain will alter things one way or the other. Inefficient speakers, and/or a noisy room will hide a lot of problems. ;) This is what some are finding out now when switching from small inefficient 87dB mains to 100dB+ mains.

You haven't discounted anything in the article, you just say a lot of things don't exist, or will never be a problem, that anyone with a little experience with a lot of different gear knows to be real issues. That is what prompted my posting here in the first place.

I fail to see how you being able to set your system up a certain way somehow voids anything I have said, or proves your views right in all (or even most) situations. I have a mixed pro/home setup that is dead silent with my main set of speakers, but I demo a lot of gear, and know that it is not dead silent with other speakers hooked up. It takes different amps, processors, and a completely different gain setup to get close to dead silent with say the DSL SH60s in my room.

I going to give up now.
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