[Wayne A. Pflughaupt]

Please use this thread for any comments or discussion about my article Gain Structure for Home Theater.

Regards,
Wayne

 

[laser188139]

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

Hello Wayne,

It's great of you to write this article on such an interesting subject.

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 9dB.

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.

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., 9dB 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. 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.

I don't think a system like Audyssey's DynEQ affects the discussion significantly. As you are recommending doing these procedures with the AVR's volume set to maximum, DynEQ should be providing no boost to the signal. As it might even be providing a reduction in the sub range, if it thinks the levels are above reference, it would probably be best that it be disabled.

I also think any source level adjustment in the AVR does not affect the discussion. Assuming the user calibrates the analog source inputs to give the same volume as the digital inputs, one would hope that these will not drive the levels past the 0dBFS that was used when adjusting the gain.

I think that exhausts my suggestions.

Thanks for writing this,
Bill

 

[soho54]

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?

 

[Wayne A. Pflughaupt]

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


Thanks for the thoughtful comments, Bill. :T

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.

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.

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.

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.

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

 

[Wayne A. Pflughaupt]

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

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

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.

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.

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.

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

 

[GranteedEV]

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

So the gyst of it is...
I need to find not only a HPF and DSP for my sub, but has to be a low noise HPF and DSP?
Great.

 

[Wayne A. Pflughaupt]

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


Not really, noise from substandard components is virtually no issue with sub woofers because it is not audible. This is why you’ll hardly ever see noise specs of any kind for manufactured subs. Sorry for not clarifying that, I’ll make the necessary changes to the text.

By the way, unless you are bottoming out your sub, or driving the amp to clipping, you don't need a high pass filter.

Regards,
Wayne

 

[Sonnie]

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

Fine and outstanding article... :clap:

 

[soho54]

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.

 

[Wayne A. Pflughaupt]

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

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.

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.

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.

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:

I was considering a DCX2496 but had worries about three things. First, the pro level; second, how to get a full signal to the DCX for good digitization (i.e. keep the volume control after the digitization to avoid digitizing a signal 50 dB below max) ...

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?

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?

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

 

[soho54]

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

Let me get a little long winded here.

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.

 

[lcaillo]

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

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.

I would suggest care in how you describe this. Talking about 1 bit of resolution in analog domain is like talking about IRE in digital video. That bit is not getting lost as resolution, but the dynamic range is diminished by the equivalent effective range. It is picking nits, but Wayne's goal here is to have a reference document that is factually correct and your posts are surely a big help in the constant process of revision and better targetting the discussion. It merits being correct and consistent in the application of terminology.

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.

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.

Remember that most of the people who will be making use of the document are not professionals but HT enthusiasts who need to better understand how to intgegrate consumer and pro equipment. The reason that this is all quite important is precisely that most do not understand gain and its relationship to SNR.

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. In my area of greater interest, video calibration, we run into similar issues all the time. No professional nor professional monitor would ever be set up to crush whites nor blacks, and the concept of dynamic range within the capability of the device is well understood. When we cross over to consumer products that are set up from the factory to "clip" all the time, it becomes difficult to educate the masses in a clear manner. The issue of digital vs analog terminology and differences in their signal levels that I mentioned above is a constant example of the confusion that can occur.

 

[Wayne A. Pflughaupt]

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,

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.

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.

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

 

[Man with a Flan]

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

Hi Wayne,

Let me first say thank you for your expansive treatment of this consumer/pro line interface issue. I have been trying to use my Crown K1 pro amp for two weeks now, and couldn't find much information into finding this article a few days ago. I really appreciate your enthusiasm and thorough explanations!

I have a 7.1 system with a Yamaha HTR-6160 AVR at the helm. It is rated at 110 watts/channel with all 7 channels running, although I don't believe that. I recently upgraded my fronts to Polk RTA 15tl's (91db 8 ohm 250 watt rms, four 6.5" drivers) and decided it was a great opportunity to outsource their amplification to a 350 watt/channelCrown K1 I had sitting around. I wanted more heft behind the speakers, and also wanted to ease the burden on my AVR. Initially I simply used an rca-to-1/4" cable from my pre-outs to the Crown, but I couldn't get the same volume that I got from the Yamaha. Today I realized that the 1/4" plug was not a balanced TRS but the other kind (TR?) So perhaps I should get another cord? I didn't realize this though until today. FYI The Crown has an input sensitivity of 1.4 vrms and the Yammie (according to the manual) passes a 1 volt signal from the pre-outs.

I ended up getting a Samson S-Convert, ran XLRs into the Crown, and it was much, much better. The bass was awakened compared to when the Yamaha drove them. I did some guessing here and there, though, since now I had three knobs to affect volume (gain knobs on the Crown, gain knob on the S-Convert, and of course the AVR's control. My Yamaha's volume spread is -80 -- +16.5. When I calibrated my speakers for 7.1 a while back with an SPL, I used 75db as my reference. I rarely go past -18 on the Yamaha, and my front speaker levels are set at 4.5 on a scale 1 -10. I like not maxing out my system in any setting, so I left everything alone when testing out the S-Convert. Just going by my ears, I noticed a lot more hiss during the music's quiet passages when I maxed out the sensitivity gain on the amp. And when instead I maxed out the S-Convert's gain and moved the Crown's knob back to its middle setting I could hardly hear any noise. The AVR was also playing a lot louder than normal at -11. I was pretty impressed with the results, but still felt there were other ways to measure than my ears and my gut. The next day I found your article.

I decided to do the voltage test on my Yamaha pre-outs. I got a digital multi-reader and downloaded your sine waves and pink noise.

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.

Next thing I did was reconnect the pre-outs to the S-Convert and take a voltage reading from the balanced XLR cable that's headed to the Crown. The S-Convert was at it's middle setting on its gain knob, which the manufacturer refers to as its "unity gain". This time I got to -12 on the Yamaha, also just over 2 volts on the multimeter, before the receiver turned itself off.

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. I guess that wave is so pure it just sounded extra loud.

So I'm kind of at a stand still because:

1) After my amp shutting down so much I'm concerned about busting speakers when the 1 khz distorts through my tweeters.

2 I don't understand why my voltage test shut the AVR down. BTW Just out of curiosity I tested the voltage on the back of the DVD player, also with the 60 hrz track. Both with the analog rca and the coax digital outs the signal was 4v, and in fact even more because it was going immediately to error on the multimeter since I was measuring in the 4v range. I just checked the Yamaha manual and noticed a "maximum input voltage (effect on, 1khz, 0.5% THD)" = 2.3v. I 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?

3) I'm also wondering if I should return my L and R speaker levels to it's SPL calibrated levels on my Yamaha, put the volume to -20 (a median movie watching level for me) and see what voltage it's delivering to the S-convert, and concurrently the amp.

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

5)There also is another piece of hardware called the Ebtech Line Level Shifter. It does the same as the S-Convert except it's passive circuitry, and requires no power. So theoretically it is sonically transparent and conveys no noise floor of its own?

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

I hate to overthink all this considering it sounded good, but in respect to all the data you provided, I'd really rather know the signal chain is running clean and trouble-free.

Sorry for trying to write more words than your article had! This is my first post, but I plan on easier topics after this! Like, "Name your favorite subwoofer movie?"

Thanks!

 

[Man with a Flan]

Bump?

 

[soho54]

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.

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.

 

[soho54]

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.

 

[Wayne A. Pflughaupt]

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

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

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?

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.

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

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?

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.

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

 

[Wayne A. Pflughaupt]

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

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.

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

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.

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

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.

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.

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

 

[soho54]

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.