Gain Structure for Home Theater Discussion Thread

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


Thanks for the thoughtful comments, Bill. :T

laser188139 said:

You might want to add to this what the impact is of using an AVR with a built-in equalization system such as Audyssey. I've thought about this a little in terms of my Denon, even though I use external amplification only for the powered sub. It strikes me that the effects come from two areas that affect the point at which clipping might occur: the AVR may have different level settings for each speaker, and the static equalization may boost certain frequencies up to 12dB.

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In Part 9, where you discuss how much headroom to allow, it would be nice to allow enough headroom for the maximum boost that the equalization system might introduce, e.g., 12dB for Audyssey. This would ensure that no clipping would occur from the voltage peak. The speaker trim might come into play here, too. Obviously if one has verified that clipping does not occur with the trims set to 0dB, and if the AVRs calibration sets a higher trim to achieve reference levels, clipping might now appear.

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Yeah, good point about Audyssey. Didn’t even consider it, because I don’t use it. But I don’t really see any issues with it WRT my home-grown gain-setting process. Maximum clean pre amp output is the maximum usable output, whether or not Audyssey’s EQ is there or not. I assume that the AVR manufacturers that incorporate Audyssey into their receivers have taken care of all that. I did note that if people were concerned about pre amp headroom when setting the amp gains, they can use a lower AVR volume setting than max. Seems to me that should be sufficient. :T

EDIT: Part 9 has been re-worked to cover Audyssey and other auto-EQ features, as well as outboard equalization.

In Part 7, where you discuss measuring the maximum voltage from the AVR, the setting of the speaker trim might matter, especially if it was very negative. For purposes of the experiment, it might be best to set it to at least 0dB, or perhaps to its maximum positive value.

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Right. And I did recommend setting speaker levels to their highest setting in the 3rd paragraph under the “How to determine your AVR’s output voltage: “My AVR has all speaker-level settings referenced to the main left and right channels, which are fixed and cannot be adjusted in the menu. If your AVR allows for adjustments for the front left and right channels, they should be set to maximum for this exercise, as should the center channel and subwoofer if you intend to measure those too.”

Related to this, adjusting the amplifier's gain to the highest level that avoids clipping at maximum signal affects how the AVRs volume setting is calibrated for reference levels. At this highest possible amplifier gain, you might have a situation where the AVR cannot set the speaker trim low enough to correctly calibrate the volume level.

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I expect this would only be an issue if someone was mixing high- and low-efficiency speakers in their system. It would be the efficient speakers that would be dialed back in the AVR. But since efficient speakers don’t need much power, it might not matter. If it did – dialing the AVR speaker levels back would mean the amp gains could be ratcheted up to compensate. Alternately, the speaker adjustments could just be performed via the amp gains instead of the AVR menu.


Regards,
Wayne

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

soho54 said:

I missed the documentation. Where are the distortion, noise floor, and headroom graphs, and information to back this method up?

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Relevant graphs were presented in Parts Three and Five. Perhaps not everything you're looking for was included, but it's far more graphs and documentation than you'll find in the typical PA system gain structure article.

I don’t have a problem answering some of the question you edited out of your post. I don’t think this topic as it relates to home theater has ever examined in depth before, so it’s natural that people will have questions and concerns. I tried to be thorough, but I doubt I thought of everything.

It is also possible/common to get higher than what should be 0dBFS out of, well pretty much everything.

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Well, it’s the maximum a DVD disc itself will put out, as it’s a digital media and will not support a higher level without distortion. I assume the DVD manufacturers are aware of that and will make sure the audio on their discs does not go beyond 0 dBFS and distort. 0dBFS 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.

It is a bit different with an AVR, of course, since the pre-amp outputs are Vrms. But it would require source component connected to the AVR with a hotter signal than 0 dBFS to get more output from it than would be provided with a DVD player or similar media component. I can’t imagine what that would be, unless someone’s plugging their Behringer XENYX mixer to their AVR. If you want to count something like that, then you’re correct: It’s possible to get higher than a 0 dBFS signal. But by and large setting amp gains with the signal generated by the 0 dBFS signal source is perfectly sufficient.

There are a couple of other explanations that seem a bit off as well, form a pro audio mindset.

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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.

This "Make sure all speaker-level settings in the AVR's menu are set to max," is just bad advice. It is the flaw behind the whole thing. An AVR like any other device has a set operational range, and headroom built in.

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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 “maximum clean 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?

Regards,
Wayne

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

soho54 said:

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.

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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. In most cases it’s possible to drive a pro amp with a consumer front end, especially if one is chosen with an ample sensitivity rating.

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.

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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.

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.

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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, such as we see here in this post from another Forum: As another 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.

Sure, it “it pays to think about the things home audio takes for granted as a given.” For instance, a couple of the things home audio has always taken for granted is quiet noise floors and not having to jack around with amp gains. IMO the main thing to be aware of is system compatibility – i.e. making sure the home and pro gear chosen is compatible. Such as not trying to use an amp with a higher sensitivity voltage than your AVR can generate. Or the possibility that cheap pro gear may add a lot of background noise. Hopefully with this piece people will now be able to determine where the weak link is in their signal chain, if there is one.

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?

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I expect people have the good sense to know if they are getting distortion and can easily tweak things to get the desired results. As I’ve noted more than once, anyone worried about the pre-outs clipping can easily use a lower setting for gain structuring.

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?

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I would hope that people would have the good sense to re-calibrate if they make equipment changes. Not sure why you expect that they wouldn't.

It’s impossible for me to conceive of or address every scenario in existence. However, following my suggestions HT enthusiasts hopefully have the tools to determine for themselves what their system needs.

EDIT: In light of your concerns about not addressing systems with low sensativity speakers etc., and Bill's comments along the same lines in his post, I've added new text to Part 9 to clarify that my suggestions are general in nature and cannot possibly address every system configuration in existance. I’ve also added text to cover gain structure with both outboard equalizers and auto-EQ systems like Audyssey. Thanks for bringing these things to my attention. :T

Regards,
Wayne

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


Yes, your comments have been very helpful, soho.

Re your concerns about the ”oddities with some devices and 0 dBFS signals.” Actually that very issue was kind of nagging at me in the back recesses of my brain when I was writing, that we probably can’t expect absolute uniformity from everything from HTIB’s and high end gear. Unfortunately it never burst through to see the light of day. :laugh:

Nevertheless, I still don’t see any reason to think we should be concerned about using a 0 dBFS signal for a gain structuring exercise. I assume that the consumer hardware manufacturers are already calibrating A/D converters with some headroom, just as the pro manufacturers do. On top of that, I’m sure that post production and software manufacturers will top out their DVDs below 0 dBFS. My main concern is that the reference signal needs to equal or exceed the maximum that can be expected from the media. It should not be appreciably lower than what’s expected from the media.

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

Unfortunately, it is true [that operating pro gear at less than +4 will cause added noise and reduced dymanic range]. 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,

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Basically you're re-stating the tired max-levels theory, which I've sufficiently debunked as bogus. 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.

It's simply not necessary for the upper signal limits of pro audio to be reached in order to obtain the best dynamic range in a home theater. Look at those graphs in Part 3 again. Consumer audio by its very nature has considerably less dynamic range than pro audio, because it's not required to deal with the crest factor of extremely high “live” signal levels. But just because the home theater signal chain does not utilize the full peak-signal capability of pro gear, that doesn’t mean we’re limiting the pro gear's dynamic range. It only means we don’t need all of it.

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.

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It sounds like you're talking about your specific equipment, which doesn't really help the rest of us. My AVR's master does not have any +dB settings, only -dB settings. If you're talking about the internal settings for the various channel's there's no universal standard as to what those figures actually reference.

This is without getting into distortion levels...

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.

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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Any measurements would only be relevant for the particular equipment being measured - i.e. a specific system. That’s not particularly helpful to the big picture.

And that’s probably why most gain structure articles don’t include measurements. They’re written to help the “man on site” perform the process without dragging along a portable laboratory of test equipment, so I don’t see why I should be held to a higher standard than paid professionals. Helping out fellow enthusiasts who aren’t especially technically-inclined, or who don’t have access to testing equipment, was the whole purpose of this piece.

There is no reason to expect that distortion will be an issue after the gain structuring exercise. As I’ve noted, no one runs their system wide open in normal use. When the pre amp is turned down to its normal operating range, there will be ample headroom and dynamic range, and the signal will be as clean as the equipment in the system allows.

However, I plan to do some additional experimenting on pre-amp distortion and will modify the thread as needed.

As Leonard mentioned, the goal is to help out all the home theater folks who have been wandering in the wilderness and getting conflicting information on gain structuring from poorly-informed sources. We’re not trying to get laboratory perfection. But naturally it would be best for the advice and suggestions to err on the side of caution.

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.


Regards,
Wayne

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

Man with a Flan said:

I rarely go past -18 on the Yamaha, and my front speaker levels are set at 4.5 on a scale 1 -10.

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That may be part of the problem. I’d suggest setting the front L/R speakers as high as you can in the menu. Naturally, the rest of the speaker levels have to be considered, but I’d raise all the speaker levels as high as possible. For instance, whichever speaker you have set highest now – say, you have the mains set for 4.5 and rears set for 7. Assuming all the speaker adjustments have the same 1-10 scale you mentioned, I’d raise the rears +3 dB to get them up to 10, and all the other speakers +3 dB as well (i.e. 4.5 becomes 7.5). Make sense? Don’t worry about “maxing out any setting.” It’s a “chicken and eggs” thing: Lower per-channel settings in the menu will mean a higher master volume setting is required in regular use, while the reverse is true for higher per-channel menu settings. In the end, what the main volume control delivers is what matters.

know I was testing with 60hz, but is it possible the Yamaha shut down because of it's input limit, not output limit? Does its volume knob simply control its own input sensitivity just like the knobs on the S-Convert and the Crown K1?

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Not likely. Typically input sensitivity controls, if they exist, are a separate adjustment. For instance, the old Yamaha DSP-A2070 5.1 integrated amp I used back in the Dolby Pro-Logic days had a menu function that could trim adjustment for each input ± 6 dB.

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.

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No, you aren’t going to damage anything – the protection mode will prevent that.

That said, I’m unsure why that’s happening. I can’t imagine how or why a receiver would go into “protect” with no speakers connected, but gear these days is a lot different (“smarter”) than it used to be. Who knows, it may be sensing a strong, relatively steady-state signal as something that would be potentially damaging to speakers (which it is!), and mandated by the legal department as something that needs to be prevented.

It may be that a dummy load is needed to “pacify” the amplifier section. Typically this is something like an 8-ohm, 100-watt resistor.

The S-Convert has a s/n ratio of 90 db
Yamaha: "s/n ratio (IHF -A Network) CD, etc (effect off, 250mv) --100db or more
Crown has >100

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The Crown’s noise spec is also A-weighted, which is a disappointment. Still, in the end all that matters if noise is at what you’d consider an acceptable level.

4) Maybe I simply need to get a rca-to-TRS interconnect? It will balance the signal? But not boost it?

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No, it takes a passive transformer or active circuit to convert unbalanced to balanced. It can’t be done with a cable.

The next step was running the 1 khz sine wave through a speaker. I had to go through the S-Convert but left it at unity gain, and the Crown's sensitivity gain was just at the 2nd notch, the first notch that made the sine wave audible. I got to -34 on the AVR and just quit. It was loud enough to hurt my ears, but no harmonic overtone.

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I assume you were trying to do the “clean voltage” test. The S-convert could be used to reduce the signal to the amplifier further (beyond what you can get with just the amp’s gain controls), so that the speakers wouldn’t be playing so loud. This will have no effect on the “clean output” measurement you’re after.

The maximum- and clean-voltage tests are mainly “FYI” exercises. It’s useful information to have before you go shopping for an amp, to make sure you don’t waste your money on one your receiver can’t drive. Since you already had your amp, you can just cut to the gain-matching exercise described in Part 8, using the pink noise and SPL meter - i.e. comparing one channel that’s receiver -> speaker to the other channel that’s receiver -> amplifier -> speaker. If you can get a gain match, you’re good to go (don’t forget to first push your per-channel settings in the menu as high as possible). If not, and the differential with the receiver -> amplifier -> speaker is unacceptably low, then use your S-convert at its lowest gain setting that will achieve the gain match.

Let us know how things go. :T

Regards,
Wayne

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

soho54 said:

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.

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You are confusing SNR (signal to noise ratio) with dynamic range.

soho54 said:

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.

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Nope. The cause of the hiss is components in the signal chain with high background noise levels - IOW, poor SNR.

soho54 said:

This is worded a little oddly. The component noise floor is static, but the component SNR and the total system noise floor is not.
<snip>
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.

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How do you propose we raise the input voltage? This is home audio, not pro audio. We don’t have the pro audio luxury of input gain controls to improve the S/N of the signal source. Instead, our front-end signal levels are fixed by the source components and AVR. The only way to increase the signal level post-AVR is to use an external device, which will also boost any background noise from the AVR by whatever amount you boost the signal. Kinda nukes the increased S/N we’re after right out of the hole.

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.

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Exactly how and why is the signal automatically “amplified” as it passes from one component to the next? It won’t be automatically amplified, and it most certainly should not be manually amplified. That would be contrary to standard gain structure protocol. According to most professional references, post-pre amp signal boosting is not recommended because it will increase the noise floor from the source component and pre amp. This is supported by the Rane article referred to in Part 2 (among other sources), which notes that the only gain changes that should be effected in downstream processors is to counter what might come from the processor itself – like an overall change in signal strength from an equalizer, for instance. The signal is not – and indeed should not – be “amplified” from one component to the next. I think you’d be hard pressed to come up with any professional references that say otherwise.

This was all discussed - and adequately discounted - throughout the article. The meters on my pro-audio Yamaha EQs rarely get above -30 to -24 (i.e. < half-scale) and indeed most of the time they’re barely getting off the bottom. Yet my system is dead silent. :whistling:

Quiet (read quality) equipment is what determines system noise, not signal levels. Every gain structure-related thread I’ve ever seen that dealt with a noise issue, the problem was ultimately isolated to a certain piece of equipment (a classic example can be seen in the link to an AVS thread found in Part 4). I’ve yet to see a thread where a noise issue was determined to be caused by inappropriate signal levels.

You’ve spent two pages now rehashing the same arguments, but have yet to offer a better solution or procedure for doing this. :huh:

Regards,
Wayne

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

soho54 said:

What I posted is fact, and can be easily demonstrated. The input and output levels at every stop along the chain will effect the total/final/in room system noise floor.

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This problem could easily be avoided by making sure the input and output levels are the same for each processor in the signal chain – as they should be. Standard pro-audio gain structure protocol, that in this case works for home theater too.

soho54 said:

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.

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If your system is dead silent and you get noise after changing out a piece of equipment, it should be obvious that it is an inferior piece of gear.

soho54 said:

I going to give up now.

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We are still waiting for your recommendation for a better method for “regular Joes” who don't have any professional calibration equipment.

dsime42 said:

I'm just a little confused here.
The "post-pre amp signal boosting is not recommended" statement seems diametrically opposed to your Figure 2 where 3dB is added between the Notch Filter and again before the Limiter...

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Yes, they do seem contradictory. The graph came from a different reference source than the Rane article. As I noted in Part 1 a few paragraphs above the graphs, it’s not uncommon for different gain structure references to have a certain amount of inconsistencies from one to the next. For a mixed home/pro system, the Rane information is what's relevant, because we don’t use outboard components like notch filters and limiters that have their own gain-boosting capabilities (the exception being sophisticated systems that use electronic crossovers or digital speaker processors).

...but I don't see anything on that graphic labeled "preamp" hence my confusion.

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Keep in mind that the graphs depict a professional PA system. In a PA system, the mixing console has essentially the same function as a home audio pre amp – i.e. it’s the piece that all the input sources plug into.

When I look at Figure 1 I see a System Dynamic Range (SDR) of 72dB and the Figure 2, after alignment, shows an SDR of 90 dB. Then in the rest of the article you make the case that the original 72dB was enough and in fact loss of an additional 6 or 12 dB would not be an issue. Sorry I think I missed something.

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Sorry but I have no idea how you arrived at the conclusion that I said the original 72 dB SDR was adequate. The primary purpose for those two graphs was to show that a proper gain structure would improve system dynamic range. The equipment depicted in the graphs is pro gear from top to bottom, and as such isn’t specifically relevant to a mixed home/pro system. The only reference I made to a “loss” of signal not mattering was specifically referring to any 24-bit digital processors used in a home system downstream from the pre amp.

Regards,
Wayne

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


Thanks for the kind words, Devon. :T

DevonD said:

The Peavey owner’s manual recommends keeping the gain dial turned all the way clockwise for “maximum headroom” and this is how I have always ran it. I have noticed that at lower volumes the subwoofer doesn’t seem to do much, I have felt the volume the speakers turned down and the volume the subwoofer turned down wasn’t parallel to each other and there was a certain point the subwoofer finally would kick in or drop off.

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The information from the Peavey manual doesn’t make much sense. Any amp naturally gets its highest noise floor with the gains all the way up, so you actually get more dynamic range (defined as the “distance” between the noise floor and highest clean input signal) with the gain reduced. Of course, this assumes that the incoming signal is hot enough to begin with to accomodate lower gain settings.

As far as the inconsistent levels with various volume controls, I have no explanation for that. It sounds like some kind of signal gate is in play. If you don’t know what that is, it’s an effect commonly used in pro audio to shut the output signal down when the input signal falls below a certain threshold. For things like drum mics, for instance, to keep cymbals from bleeding into the snare drum mic when it’s not being hit. Check the miniDSP to see if has some kind of feature like this, and if so disable it.

In case you haven’t already, I’d set the Peavey’s back-panel filters for full range and let the AVR or miniDSP provide the low pass filtering.

I am under the impression, and hope, that the MiniDSP outputs a signal equal to what it is receiving so is there a need for me to optimize the gain structure on the MiniDSP? And if so how would I perform this? One idea I had was to adjust the gain from the receiver to the amplifier just to the point of the clip light flickering. Then adding the minidsp in between the two and getting it to the same point where the clip light on the amp starts to flicker. Would this work well enough?

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Like any other outboard equalizer, the miniDSP should not need any level settings to optimize gain structure. It should pass the signal straight through (along with whatever is added with any equalization, of course). So what you’d want to do is re-adjust the Peavey’s gain settings after the miniDSP is added to the signal chain, to accommodate any changes to gain the equalization brings.

My only concern is that the miniDSP, with only 2 Vrms capability, it might not have enough headroom to pass the signal level the Denon as actually capable of generating, and will clip. As noted in Part 7, the receiver I used while researching the article put out way more voltage than the manufacture claimed, so I wouldn’t put much stock in Denon’s spec.

One more question. If, for example, I end up finding that having the subwoofer gain in the AVR at +6 and the gain on the amplifier at say -12 is the optimal level to have their gain matched what should I do when I run audessey and it turns the subwoofer down to -3 or -1 or whatever it may be. Should I turn the gain on the amplifier up the level that the receiver was turned down?

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If you’re not happy with the sub level Audyssey generates, then I’d over-ride it and set it where you think it should be. If Audyssey won’t let you do that from the receiver specifically for the sub output, then it’s easy enough to change it on the back end – i.e., the amplifier gain controls.

Hope this helps!

Regards,
Wayne