# DCX : How to create an LT? - Is it the right device?



## Drizt

Hi Guys,

Ive had a bit of a play around with the software and got a bit of a feel for it, but wanted to ask those that actually used the device before if its a good thing to use to create an LT?

Also if you wouldn't mind could you add a run down on what you did..

Cheers.


----------



## Drizt

Also, what are the negatives of the unit?

Does it cause a thump at turn on?


----------



## Wayne A. Pflughaupt

What is a LT?

Regards,
Wayne


----------



## Drizt

Linkwitz Transform -> http://sound.westhost.com/linkwitz-transform.htm

Heres a calculator from this site -> http://www.hometheatershack.com/forums/diy-subwoofers/9067-linkwitz-transform-calculator.html


----------



## Drizt

Drizt said:


> Also, what are the negatives of the unit?


I know the unit outputs pro levels, which I believe my Quest 3004 can handle (theres a button for lowering the gain - I think thats all that is required).

And I believe the unit works better with pro levels on the inputs? Is that correct.

The Behringer MX882 is what Im planning to use to Sum the LFE + L & LFE + R and possibly up the inputs to the DCX to pro levels.

Just wanting to be over cautious and ask as many questions as I can before I jump in and possibly buy the wrong combination of kit.

Any thoughts?


----------



## lsiberian

The DCX can do anything eq wise that can be done so yes it's the right device.

To use it you will need an amp with trims or XLR-RCA with -12db attenuators.


----------



## Drizt

lsiberian said:


> The DCX can do anything eq wise that can be done so yes it's the right device.
> 
> To use it you will need an amp with trims or XLR-RCA with -12db attenuators.



Thanks for the reply.

I have been looking through the Quest 3004 documentation and have found the following.

There is input trims on the front and there is a gain selector on the back.



> amplifier performance
> • Amplifier gain
> The amplifier should always be operated with the input
> level controls set at maximum. The only departure from
> this practice is where the internal filters/crossover is
> engaged in a bi-amp configuration, where it may be
> necessary to turn down the high frequency side of
> the amplifier. If the amplifier needs to be calibrated, to
> match a mixer line level output, the gain switch on the
> back of the amplifier can be switched to the correct
> line level.


On the back of the unit the gain setting options are:



> 7. Selectable Gain Settings
> Adjustable input switching for 1.4V, 32 dB or 26 dB make system calibration easy.


What option should I be using given the output of the DCX ? What is the 1.4V setting and how does it compare to the other two options ?


----------



## Drizt

Can anyone confirm ?



Drizt said:


> Thanks for the reply.
> 
> I have been looking through the Quest 3004 documentation and have found the following.
> 
> There is input trims on the front and there is a gain selector on the back.
> 
> 
> 
> On the back of the unit the gain setting options are:
> 
> 
> 
> What option should I be using given the output of the DCX ? What is the 1.4V setting and how does it compare to the other two options ?


----------



## brucek

> 7. Selectable Gain Settings
> Adjustable input switching for 1.4V, 32 dB or 26 dB


The 26dB and 32dB settings are fixed gain settings that assume the pro standard input. So, for an input level of 0dBu (=0.775vRMS), the output of the amp will produce a voltage gain of 26dB or 32dB depending on the switch setting.

The 1.4V setting is a sensitivity setting (as opposed to fixed gain) that says the amplifier will produce its full power output (i.e. 1100watts @4ohms) with 1.4volts input. The 1.4v is a broadcast standard where 1.4v=0VU.

The DCX output will be at the standard pro level, so I would choose either 26dB or 32dB. If 32dB peaks your limiters, then drop to 26dB.

The MX822 can also set any level you wish, so I don't see that you have any problems. It's all balanced and all at pro levels.

brucek


----------



## Drizt

Thank you.



brucek said:


> The 26dB and 32dB settings are fixed gain settings that assume the pro standard input. So, for an input level of 0dBu (=0.775vRMS), the output of the amp will produce a voltage gain of 26dB or 32dB depending on the switch setting.
> 
> The 1.4V setting is a sensitivity setting (as opposed to fixed gain) that says the amplifier will produce its full power output (i.e. 1100watts @4ohms) with 1.4volts input. The 1.4v is a broadcast standard where 1.4v=0VU.
> 
> The DCX output will be at the standard pro level, so I would choose either 26dB or 32dB. If 32dB peaks your limiters, then drop to 26dB.
> 
> The MX822 can also set any level you wish, so I don't see that you have any problems. It's all balanced and all at pro levels.
> 
> brucek


----------



## Drizt

Found a good post on how to create LT's using the DCX (and other such devices)

http://www.htguide.com/forum/showpost.php4?p=125637&postcount=3



> Most of the better digital EQ boxes, e.g. the Behringer DEQ2496 or DCX2496 can do it with ease.
> 
> The LT can be replicated as the sum of 3 filters - two band (bell shaped) filters and one 12dB/octave shelving filter. The first band filter changes the Q of the box/driver to 0.7 at it's Fb. The shelving filter changes the response to Q=0.7 at the new desired Fb. The second band filter sets the Q to whatever you want it to be, e.g. Q=0.5, at the new Fb.
> 
> In practice, you may not even need the band filters. Just use a shelving filter to boost the lows. If the final response curve looks the way you want it to, it doesn't matter how you get there or what kind of fancy name you give it. It's all just EQing the bass response to be the way you want it.


----------



## Drizt

I ended up sending an email to Quest to see what setting they recommend, and they said to use the 1.4V setting. 

, I wish I knew more than mooo all about electronics 



brucek said:


> The 26dB and 32dB settings are fixed gain settings that assume the pro standard input. So, for an input level of 0dBu (=0.775vRMS), the output of the amp will produce a voltage gain of 26dB or 32dB depending on the switch setting.
> 
> The 1.4V setting is a sensitivity setting (as opposed to fixed gain) that says the amplifier will produce its full power output (i.e. 1100watts @4ohms) with 1.4volts input. The 1.4v is a broadcast standard where 1.4v=0VU.
> 
> The DCX output will be at the standard pro level, so I would choose either 26dB or 32dB. If 32dB peaks your limiters, then drop to 26dB.
> 
> The MX822 can also set any level you wish, so I don't see that you have any problems. It's all balanced and all at pro levels.
> 
> brucek


----------



## Drizt

I found a very helpful post that explains the math behind the decision to use the 1.4V setting for gain on the Quest 3004 (in my case).

http://www.controlbooth.com/forums/sound/8797-use-plus-4-minus-10-4-10-switches.html#post102969

Lets hope its right :clap:



> Re: Use of plus 4/minus 10 (+4/-10) switches?
> Almost.
> 
> I was working through the math of this the other night for a project. +4dBu is a voltage signal, referenced to 0.775 V. Thus, 4 dBu is actually 1.228V. (4 = 20*log(V/0.775); thus 0.2 = log(V/.775) and V = 0.775*(10^0.2))
> 
> -10 dBV is also a voltage signal, but referenced to 1V. So this signal would be 316 mV (-10 = 20*log(V/1); thus -0.5 = log(V) and V = 10^-0.5 = 0.316).
> 
> Thus, the difference in voltage would be: V = 20*log(1.228/0.316) = 11.79, or approximately 12 dB difference in signal strength.
> 
> THe short answer to your question is that almost all consoles reference their outputs to 4dBu, unless noted. Most IEM gear will probably have a switch on it. So just set the switch to +4 dBu and plug it up.
> __________________
> Mike Benonis
> Grad Electrical Engineering '14, Virginia Tech
> Electrical Engineering '09, The University of Virginia
> KI4RIX
> http://www.benonis.net/


----------



## brucek

> explains the math behind the decision to use the 1.4V setting for gain on the Quest 3004 (in my case).


I don't see how this encourages you to use the 1.4 volt setting?

The DCX has a max output pro level of +22dBu = 9.75 VRMS. math = (0.775 antilog (22 / 20).

So if the amp setting of 1.4 says that full power output is reached when 1.4 volts is at the input, what will happen when you feed the amp 9.75 volts (+22 dBu)?

brucek


----------



## Drizt

To be honest its from piecing all the information together and trying to join the dots together. I admit I know very little about electronics. 

I have been told in this thread that DCX would out pro levels and that to bring it down to consumer levels it would need to be attenuated by 12dB. The math above, I thought, showed that by using the 1.4V sensitivity would in effect account for this 12dB difference. It also shows that at pro levels (+4dBu) that the output would be below 1.4V, so never able to achieve full output.

Also I emailed quest about using pro level inputs into the amp and they said to use the 1.4V setting.

The figure of +22dBu you list below is the first I have seen in reference to the DCX. Can you link to this for me please?





brucek said:


> I don't see how this encourages you to use the 1.4 volt setting?
> 
> The DCX has a max output pro level of +22dBu = 9.75 VRMS. math = (0.775 antilog (22 / 20).
> 
> So if the amp setting of 1.4 says that full power output is reached when 1.4 volts is at the input, what will happen when you feed the amp 9.75 volts (+22 dBu)?
> 
> brucek


----------



## Drizt

brucek said:


> The MX822 can also set any level you wish, so I don't see that you have any problems. It's all balanced and all at pro levels.
> 
> brucek


Ill be using the MX882 to mix channels on the input side to the DCX. I had a look at feeding the outputs of the DCX back into the MX882 but from what I could see that would end up creating a feedback loop if I used the main outputs which I intend to.


----------



## brucek

> I have been told in this thread that DCX would out pro levels and that to bring it down to consumer levels it would need to be attenuated by 12dB.


No, you will be using all pro levels and all balanced. There is no need to convert to consumer level. No attenuation required.



> It also shows that at pro levels (+4dBu) that the output would be below 1.4V, so never able to achieve full output.


No, the +4dBu refers to the nominal pro level, just as -10dBv refers to nominal consumer level. The max levels are listed in the device manuals. 

The MX882 spec sheet (download here), and the DCX is the same (download here). Both +22dBu max out.

brucek


----------



## Drizt

How have you come to that? Im lost.

I will be using the Integra DHC-9.9 which has XLR outputs. But I assume this is a consumer level product and will not output pro levels?

I may also integrate my 2ch rig with the HT rig and would then need to mix single ended (RCA converted to XLR) with the LFE from HT via the MX882.

The DCX works best a pro levels so I could use the MX882 to raise the levels to pro levels.

The quest 3004, i have ****. I had assumed if i feed it the pro levels from the DCX that it would have a fit. And that I would need to compensate for the 'pro' levels.



No wonder people pay extra to just buy consumer level goods that do it all for you. Too much stuff to work out for most people I would imagine.

I must say you have thrown a spanner in the works. Your telling me different information than what the equipement manufacturer is telling me.



brucek said:


> No, you will be using all pro levels and all balanced. There is no need to convert to consumer level. No attenuation required.
> 
> 
> No, the +4dBu refers to the nominal pro level, just as -10dBv refers to nominal consumer level. The max levels are listed in the device manuals.
> 
> The MX882 spec sheet (download here), and the DCX is the same (download here). Both +22dBu max out.
> 
> brucek


----------



## brucek

> How have you come to that? Im lost.


The MX882 can set any level you want. You want pro.



> I had assumed if i feed it the pro levels from the DCX that it would have a fit.


It's a pro level amp. I see no reason for it to have a fit if you send it pro levels. They provide a sensitivity and fixed gain switch along with a GAIN Input switch to calibrate mixer line levels to amplifier input sensitivity.



> No wonder people pay extra to just buy consumer level goods that do it all for you. Too much stuff to work out for most people I would imagine.


The pro equipment gives you options.



> I must say you have thrown a spanner in the works.


You may be worrying about it too much. If one setting doesn't work, try another.

brucek


----------



## Drizt

Thanks for your help.

I guess Im just confused as the Quest guys told me to use the 1.4V sensitivity setting. 



brucek said:


> The MX882 can set any level you want. You want pro.
> 
> 
> It's a pro level amp. I see no reason for it to have a fit if you send it pro levels. They provide a sensitivity and fixed gain switch along with a GAIN Input switch to calibrate mixer line levels to amplifier input sensitivity.
> 
> 
> The pro equipment gives you options.
> 
> 
> You may be worrying about it too much. If one setting doesn't work, try another.
> 
> brucek


----------



## Drizt

brucek said:


> You may be worrying about it too much. If one setting doesn't work, try another.
> 
> brucek


If I try a setting which causes the input on the Quest to 'clip' could that damage a subwoofer driver? I know clipping can kill a tweeter pretty quick, but what about a subwoofer?

I found a nice simple PDF to explain the fixed gain and sensitivity settings on amps in case anyone is interested -> http://www.proaudiosystems.co.uk/do...ures relevant to loudspeaker controllers..pdf

I think now I understand what brucek is saying about causing clipping if using the 1.4V setting.

Is this right (I posted this elsewhere to):

Scenario 1 (NO EQ):
If we gave normal pro levels (+4dBu) to the DCX and applied NO eq, then the DCX would output pro levels (+4dBu) which would mean selecting the 1.4V sensitivity gain setting on the Quest 3004. This would mean that +4dBu would be the highest output coming from the DCX (representing 100% signal). Is that right?

Scenario 2 (With EQ):
Lets say I use 6dB gain at 20Hz to extend the bass output of my sealed Maelstrom-X subs. With the same pro levels in, would the DCX now output +10dBu (+4dBu + 6dB gain) as its maximum output for a 100% signal strength for a 20Hz tone ? So if the Quest was set to 1.4v and it recieved a +10dBu signal it would cause clipping ? 



So in this case if using the 1.4V sensitivity setting on the Quest, would I need to apply a negative gain (attenuation) of 6dB to the inputs on the Quest to ensure no clipping occurred?



If your using the fixed gain settings, how do you know how much is too much gain?
Eg. Lets say the DCX was outputting +10dBu at 20Hz for a 100% signal strength (I hope I have the terminology right) like in scenario two, that would mean there would be 36dB gain if using the 26dB fixed gain setting or 42dB gain using the 32dB gain setting.

How much gain is to much ?


----------



## brucek

There isn't much chance in damaging a subwoofer with a clip now and then, but I don't see much chance of that happening to you with that amplifier. It has Peak Status LED's that warn you that you are approaching clipping (not clipping, but approaching clipping). Then you have a switchable Limiter control that prevents the amplifier from clipping or distorting.

You can't look at a music signal as some steady state sine wave that is at exactly +4dBu. The standard reference to +4dBu and -10dbV are nominal readings for pro and consumer. They are not peak - but more of an average. The pro output peak on your devices driving your amplifier are +22dBu. You don't need any EQ to reach those levels, but an EQ filter of +6dB will certainly raise the level by that much at that specific filter frequency. So when you ask, "_This would mean that +4dBu would be the highest output coming from the DCX (representing 100% signal). Is that right?_, the answer is no. That's a nominal level and any peak in the music can reach +22dBu before clipping the DCX.

You have to pay attention to the gain structure of your entire chain. Too much level and you'll clip, and too low a level and your S/N raises. Most pro equipment have VU's to help you set your levels. Try and set the levels so that each stage receives the maximum signal without clipping when the loudest volume listening level is used from your receiver. Do some reading about audio gain structure on the internet - it's one of the more important subjects in the pro equipment world.

brucek


----------



## Drizt

Thanks again. That was very helpful.



brucek said:


> There isn't much chance in damaging a subwoofer with a clip now and then, but I don't see much chance of that happening to you with that amplifier. It has Peak Status LED's that warn you that you are approaching clipping (not clipping, but approaching clipping). Then you have a switchable Limiter control that prevents the amplifier from clipping or distorting.
> 
> You can't look at a music signal as some steady state sine wave that is at exactly +4dBu. The standard reference to +4dBu and -10dbV are nominal readings for pro and consumer. They are not peak - but more of an average. The pro output peak on your devices driving your amplifier are +22dBu. You don't need any EQ to reach those levels, but an EQ filter of +6dB will certainly raise the level by that much at that specific filter frequency. So when you ask, "_This would mean that +4dBu would be the highest output coming from the DCX (representing 100% signal). Is that right?_, the answer is no. That's a nominal level and any peak in the music can reach +22dBu before clipping the DCX.
> 
> You have to pay attention to the gain structure of your entire chain. Too much level and you'll clip, and too low a level and your S/N raises. Most pro equipment have VU's to help you set your levels. Try and set the levels so that each stage receives the maximum signal without clipping when the loudest volume listening level is used from your receiver. Do some reading about audio gain structure on the internet - it's one of the more important subjects in the pro equipment world.
> 
> brucek


----------



## Drizt

brucek,

Thanks again for all your help. I know it must be frustrating trying to explain it to me. I appreciate the effort.


I think I can use the one MX882 to lift the level going into the DCX and then drop them back down after the DCX.

It seems that each input can be set to split or mix mode. I originally thought the whole unit was either in split more or mix mode, but seems it can be done on an individual channel basis. I know I know, RTFM. 

So I could do...

Main inputs, are the LPF (40Hz and below) output from my active main speakers.
Input 1 could be the LFE channel which gets mixed into both the L & R main outputs.
Set the main output levels appropriately for the DCX.

This could then be feed into the DCX .
The output from the DCX could then be feed back into the MX882 on channels 3 & 4.
Set those channels to 'split' mode (so that they don't get summed to the main outputs and create a feedback loop).
Set the output levels to what is appropriate for the inputs of the Quest 3004.

Work out the appropriate gain or sensitivity on the Quest.

I 'THINK' thats how it could work.


----------



## Drizt

brucek said:


> Try and set the levels so that each stage receives the maximum signal without clipping when the loudest volume listening level is used from your receiver.


My reciever is an Integra DHC-9.9.

Would you use the tone generator in the Intregra that you use to calibrate the system to reference levels (think that 85dB) ? Or whats the appropriate method to send out the loudest volume? Sorry that might be stupid question, but just wanting to double check my procedure.

I assume you can send all these signals through the behringer and quest gear without the subs connected to the amp? Just want the lights for clipping and adjust?




brucek said:


> Do some reading about audio gain structure on the internet - it's one of the more important subjects in the pro equipment world.
> 
> brucek


Will do. Ill look into it tonight. Thanks again.


----------



## Drizt

I'm pretty sure I was wrong again on the below.
:dontknow:

I pretty sure I can't use the 'sum / mix' functionality if I want to do level matching on the outputs of the DCX as it would create a feedback loop. I 'think' I was right the first time.

I 'think' I have to scrap the idea of combining the Low Passed (40Hz and below) from the L & R mains with the Home Theatre LFE channel.

Summary: 
Input 1). could be a copy of the LFE (use a splitter cable from my Integra DHC-9.9), which I put in 'mix' mode, adjust gain for DCX input, and output straight out of Output 1.

Input 2). could be a copy of the LFE (use a splitter cable from my Integra DHC-9.9), which I put in 'mix' mode, adjust gain for DCX input, and output straight out of Output 2.

Input 3). this would be the L output from the DCX. Here we would lower the voltage to the whatever is needed by the Quest 3004 for the Left channel input.

Input 4). this would be the R output from the DCX. Here we would lower the voltage to the whatever is needed by the Quest 3004 for the Right channel input.

, I hope I have it right this time.. talk about a merri-go-round in my head 




Drizt said:


> So I could do...
> 
> Main inputs, are the LPF (40Hz and below) output from my active main speakers.
> Input 1 could be the LFE channel which gets mixed into both the L & R main outputs.
> Set the main output levels appropriately for the DCX.
> 
> This could then be feed into the DCX .
> The output from the DCX could then be feed back into the MX882 on channels 3 & 4.
> Set those channels to 'split' mode (so that they don't get summed to the main outputs and create a feedback loop).
> Set the output levels to what is appropriate for the inputs of the Quest 3004.
> 
> Work out the appropriate gain or sensitivity on the Quest.
> 
> I 'THINK' thats how it could work.


----------



## brucek

> Would you use the tone generator in the Intregra that you use to calibrate the system to reference levels (think that 85dB) ? Or whats the appropriate method to send out the loudest volume? Sorry that might be stupid question, but just wanting to double check my procedure.


I admit to being confused about your connections with reference to summing and looping back? What are you exactly trying accomplish. Doesn't your Integra have a simple subwoofer out that you intend to feed to the MX and then to a DCX and then to a sub amp. You'll have to explain your goals a bit more.

Either way, since the source for all your signal comes from a receiver that has a variable volume control, you'll have to establish your personal loudest peak signal by playing music and movies (for the LFE) to get a feel for the input level to the first stage after the receiver. Play a movie that is LFE rich at the loudest level you feel would be your maximum. You would want the MX's input to be just below clipping at that point. Then you would want its output level to be just below clipping and so on to the next stage...

brucek


----------



## Drizt

brucek said:


> I admit to being confused about your connections with reference to summing and looping back? What are you exactly trying accomplish. Doesn't your Integra have a simple subwoofer out that you intend to feed to the MX and then to a DCX and then to a sub amp. You'll have to explain your goals a bit more.


My mains are full range and active. The mains have a low pass output (40Hz and below) that can be sent directly to a subwoofer. So if I wanted to maintain stereo bass I would use these outputs and combine/sum them with the LFE channel and send those combined signals through the chain. But from looking at how the sum/mixing is done in the MX it would seem I can not do the sum'ing and then feed the output of the DCX back into the MX to down the levels as that would then get sent to the sum/mixed output as well as its own output. A bit confusing to put into words, but im pretty sure it can't be done, but thats ok.



brucek said:


> Either way, since the source for all your signal comes from a receiver that has a variable volume control, you'll have to establish your personal loudest peak signal by playing music and movies (for the LFE) to get a feel for the input level to the first stage after the receiver. Play a movie that is LFE rich at the loudest level you feel would be your maximum. You would want the MX's input to be just below clipping at that point. Then you would want its output level to be just below clipping and so on to the next stage...
> 
> brucek


Cheers. Thanks for the tips.


----------



## brucek

> My mains are full range and active. The mains have a low pass output (40Hz and below) that can be sent directly to a subwoofer.


Why would you not allow your Integra to take care of bass management, and set the Integras mains output to small @40/60/80 Hz and then use the combined subwoofer output of the Integra to feed the sub(s)?



> So if I wanted to maintain stereo bass I would use these outputs and combine/sum them with the LFE channel


Explain what you mean by this? Combining the left and right channels <40Hz signal doesn't maintain stereo, it turns it into mono. Where is this LFE channel coming from that you are combining?



> looking at how the sum/mixing is done in the MX it would seem I can not do the sum'ing and then feed the output of the DCX back into the MX to down the levels as that would then get sent to the sum/mixed output as well as its own output.


You'll have to explain why you want to feed the output back to the MX?

brucek


----------



## Drizt

brucek said:


> Why would you not allow your Integra to take care of bass management, and set the Integras mains output to small @40/60/80 Hz and then use the combined subwoofer output of the Integra to feed the sub(s)?


Because then it would be mono bass. Might be a valid option, but I would want to try stereo bass first and then compare it to mono.



brucek said:


> Explain what you mean by this? Combining the left and right channels <40Hz signal doesn't maintain stereo, it turns it into mono.


Integra Left channel goes to left speaker. Left speaker has 40Hz LP output.
Integra Right channel goes to right speaker. Right speaker has 40hz LP output.

If you sum/mix 'twice' then you will have L + LFE AND R + LFE.
Essentially sum/mis the LFE with both L & R and send the results to one sub each.



brucek said:


> Where is this LFE channel coming from that you are combining?


Integra LFE output.



brucek said:


> You'll have to explain why you want to feed the output back to the MX?
> 
> brucek


Feeding the DCX output back into the MX would mean that I could attenuate the high levels going into the Quest. It sounded like using the trims on the front of the Quest was not recommended. Not sure why. Most people are telling me that I need to attenuate the outputs of the DCX as they can up to +22dBu which would clip the Quest.


----------



## Wayne A. Pflughaupt

> I 'think' I have to scrap the idea of combining the Low Passed (40Hz and below) from the L & R mains with the Home Theatre LFE channel.


Are you intending for both signals to be in use at the same time?

Regards,
Wayne


----------



## Wayne A. Pflughaupt

> Because then it would be mono bass. Might be a valid option, but I would want to try stereo bass first and then compare it to mono.


I think you could easily accomplish this with one of the Zenyx mixers (or something similar from some other manufacturer). Send the two low-passed L/R signals to two inputs, pan them left and right. Send the LFE to another input, panned straight up. Send left and right Zenyx outputs to the EQ, then EQ outs to the amp. 

Use the Zenyx’s per-channel mute buttons to send either the mono LFE signal to the subs, or the L/R stereo.

Regards,
Wayne


----------



## Drizt

Wayne A. Pflughaupt said:


> Are you intending for both signals to be in use at the same time?
> 
> Regards,
> Wayne


For home theatre usage yes. I would set the mains to large which would mean that the subs do the LFE + L (<40Hz) & LFE + R (<40Hz).



Wayne A. Pflughaupt said:


> I think you could easily accomplish this with one of the Zenyx mixers (or something similar from some other manufacturer). Send the two low-passed L/R signals to two inputs, pan them left and right. Send the LFE to another input, panned straight up. Send left and right Zenyx outputs to the EQ, then EQ outs to the amp.
> 
> Use the Zenyx’s per-channel mute buttons to send either the mono LFE signal to the subs, or the L/R stereo.
> 
> Regards,
> Wayne


I have a Zenyz 802 that I use with my EC8000 mic for taking measurements.

If I don't send the outputs of the DCX to the MX882, then I could do the same thing with the MX882. 

If its ok to use the trim inputs on the Quest, then I could just use that to attenuate the inputs as required. Its just that the manual seemed to indicate not to use the trims for this but use the gain settings on the back. Eh, guess ill just have to experiment.

So I can hook all the units up but NOT plug the subs into the Quest (so there will be no sound at all), so there will be just the flashing lights on all the units going crazy. This is how you do the gain matching yes?


----------



## brucek

> Might be a valid option, but I would want to try stereo bass first and then compare it to mono.


You'll be running mains full range that will be subject to low frequency room modes that you can't control with EQ. Then you'll be adding bass < 40Hz from your subs to that full range un-EQ'd signal already being transmitted into the room. I see lots of possibilities for peaks. Bass management is so much more efficient.

Stereo bass won't do much for you since signals <~80Hz are non-directional in a room. Mono - Stereo, matters not at low frequencies.



> If you sum/mix 'twice' then you will have L + LFE AND R + LFE.


Why do need to sum twice? Mix left, right, LFE and you're done. Feed it to the DCX.



> Feeding the DCX output back into the MX would mean that I could attenuate the high levels going into the Quest.


The MX and DCX have lots of control over the level, as does the Quest. Why would you need to feed a signal back into the MX?

brucek


----------



## Wayne A. Pflughaupt

> If I don't send the outputs of the DCX to the MX882, then I could do the same thing with the MX882.


...Except that the MX882 does not have per-input muting. You'll have to have that if you want to be able to compare stereo and mono bass. You'll also need a different mixer, as your 802 does not have per-input muting.

If not the ability to compare stereo and mono bass, what exactly is it you're trying to accomplish?

Regards,
Wayne


----------



## Drizt

Wayne A. Pflughaupt said:


> ...Except that the MX882 does not have per-input muting. You'll have to have that if you want to be able to compare stereo and mono bass. You'll also need a different mixer, as your 802 does not have per-input muting.
> 
> If not the ability to compare stereo and mono bass, what exactly is it you're trying to accomplish?
> 
> Regards,
> Wayne


The simplest way to compare mono and stereo bass would be the following.

Set the system up as I planned with stereo bass.

Then....

Set the mains to small in the integra. 
Set the Crossover to 40Hz (or whatever - but since im doing stereo bass with the subs at 40Hz Ill start with that).
Then set the mains (active speakers) to full range without the LP sub out turned on.

This achieves the same thing you are suggesting I would have thought ?

BUT... stereo bass is what I am trying to nut out first.


----------



## Drizt

brucek said:


> You'll be running mains full range that will be subject to low frequency room modes that you can't control with EQ. Then you'll be adding bass < 40Hz from your subs to that full range un-EQ'd signal already being transmitted into the room. I see lots of possibilities for peaks. Bass management is so much more efficient.


Possibly. I have Audyssey in the Integra I could apply if feel the need. I don't really like 'black box' eq like you find in the Integra. It doesn't tell you what it is doing EQ wise. I find that annoying.



brucek said:


> Stereo bass won't do much for you since signals <~80Hz are non-directional in a room. Mono - Stereo, matters not at low frequencies.


Debatable. Don't know who is right, but I'd like to find out for myself.



brucek said:


> Why do need to sum twice? Mix left, right, LFE and you're done. Feed it to the DCX.


Maybe my explanations were poor. My apologies. Here is a diagram to help you out.










0). PS3 slim (HDMI) to Integra DHC-9.9
1). Digital link from the Duet to the Integra DHC-9.9
2 - a&b). Analogue outputs to the active SGR MT3FSL's.
3). LFE analogue output to the Behringer MX882
4 - a&b). If I choose to, I can enable the sub out on the MT3FSL's (40Hz LPF) to ultimately send to the subs for 2ch use by mixing them in via the Behringer MX882.
5a). The Behringer MX882 will sum the LFE + L and raise the output to pro levels before sending it to the Behringer DCX.
5b). The Behringer MX882 will sum the LFE + R and raise the output to pro levels before sending it to the Behringer DCX.
6 - a&b). The Behringer DCX will be used to EQ the subs separately and send the signals to the Quest 3004 (set the gain switch on the 3004 to 1.4V - pro levels) EDIT: I have some confusion over what setting to use now sad.gif
7 - a&b). The Quest 3004 drives the two SGR made sealed, passive 125L Maelstrom-X's 



brucek said:


> The MX and DCX have lots of control over the level, as does the Quest. Why would you need to feed a signal back into the MX?


Perhaps I do not need to.


----------



## Mika75

Can i suggest the following.... 

Integra --> full range signal analogue out to DCX --> DCX now controls front L/R & both Subs in Stereo or Mono (this also leaves room for a center channel input/output in the future).


----------



## Drizt

Mika75 said:


> Can i suggest the following....
> 
> Integra --> full range signal analogue out to DCX --> DCX now controls front L/R & both Subs in Stereo or Mono (this also leaves room for a center channel input/output in the future).


Its possible yes. But was only really wanting to use the DCX for the subs for now. You know the usual audiophile thinking of keeping the EQ away from the mains  Its something I could experiment with later though.


----------



## brucek

> the usual audiophile thinking of keeping the EQ away from the mains


Absolutely. 



> Here is a diagram to help you out.


OK, I don't see any reason why that won't work. No need for any loopback? The MX882 seems capable of mixing the LFE with the two main channels of low frequency info from your active speakers.

brucek


----------



## Wayne A. Pflughaupt

This exactly how I figured your connection scheme was - thanks for saving me the trouble of doing up a diagram. :T 

Ditch the MX822 and substitute a small "project mixer" that has per-input mute buttons and balanced outputs. It'll do everything you want - sum LFE with right and left (3 + 4a &4b) , or let you hear just the LFE mono, or the just the left/right in stereo. All quickly and easily accomplished with the mute buttons. Just be sure that the three (3, 4a, 4b) are all level-matched via the mixer's gain and level controls. 

If this just for an experiment, keep the receipt and return the mixer once you've made your evaluation.

Regards,
Wayne


----------



## Drizt

brucek said:


> OK, I don't see any reason why that won't work. No need for any loopback? The MX882 seems capable of mixing the LFE with the two main channels of low frequency info from your active speakers.
> 
> brucek


Yeap, i was only worried about attenuated the voltage coming out of the DCX before its input into the Quest. But I suppose I could use the trims on the quest for this. Its just that the manual seemed to indicate not to use the trims but to use the gain switch on the back. But seems I am easily confused as the DCX would be outputting much higher voltage than regular if using EQ which could lead to clipping. Sorry the whole gain matching thing is probably not going to sink in until I give it a go.


----------



## Drizt

Wayne A. Pflughaupt said:


> This exactly how I figured your connection scheme was - thanks for saving me the trouble of doing up a diagram. :T


No problems 



Wayne A. Pflughaupt said:


> Ditch the MX822 and substitute a small "project mixer" that has per-input mute buttons and balanced outputs. It'll do everything you want - sum LFE with right and left (3 + 4a &4b) , or let you hear just the LFE mono, or the just the left/right in stereo. All quickly and easily accomplished with the mute buttons. Just be sure that the three (3, 4a, 4b) are all level-matched via the mixer's gain and level controls.
> 
> If this just for an experiment, keep the receipt and return the mixer once you've made your evaluation.
> 
> Regards,
> Wayne


Already ordered the MX882.

And don't really need the muting functionality to do what I want.


----------



## brucek

> i was only worried about attenuated the voltage coming out of the DCX


Just turn its output down to the level you require. I don't understand your worry?

brucek


----------



## Drizt

brucek said:


> Just turn its output down to the level you require. I don't understand your worry?
> 
> brucek


Because everyone gives you a different answer. I guess that can be the pitfall of asking too many people the same question, you'll end up with too many different answers  

People have said to turn down the gain 'after' the DCX. I believe it was said that lowering the gain too much in the digital domain would bring the noise floor closer to the signal (lower signal to noise ration). They say to use an analogue device to match the gains between the DCX and the Quest.


----------



## brucek

> They say to use an analogue device to match the gains between the DCX and the Quest.


Utter nonsense........


----------



## Drizt

brucek said:


> Utter nonsense........


Fair enough.

What would be your preference for attenuating the gain between the DCX and the Quest.

The analog trims on the Quest input or the digital gain on the output of the DCX ?


----------



## brucek

> What would be your preference for attenuating the gain between the DCX and the Quest.
> 
> The analog trims on the Quest input or the digital gain on the output of the DCX ?


I would set all the inputs and outputs of the devices in my chain with the input and output level controls provided to obtain the best gain structure possible.

brucek


----------



## Drizt

brucek said:


> I would set all the inputs and outputs of the devices in my chain with the input and output level controls provided to obtain the best gain structure possible.
> 
> brucek


Seems like a sensible answer.

Guess I just have to have a play and find out what works and what doesn't.

I can see the appeal of a consumer grade 'black box' approach now


----------



## Mika75

Mika75 said:


> Can i suggest the following....
> 
> Integra --> full range signal analogue out to DCX --> DCX now controls front L/R & both Subs in Stereo or Mono (this also leaves room for a center channel input/output in the future).





Drizt said:


> Its possible yes. But was only really wanting to use the DCX for the subs for now. You know the usual audiophile thinking of keeping the EQ away from the mains  Its something I could experiment with later though.


The system set-up i posted is by far the most transparent and powerful, i don't believe in mixing the LFE with L/R sub bass signals. 

Most Speakers have built in crossovers, this is what u would refer to as a 'fixed EQ'

Your Active Mains have fixed crossovers, but also adjustable shelving filters for low/high (u should be using these)... this is all still EQ.

Thinking as an Audiophile, u now have some issues.. onder:


----------



## Drizt

Mika75 said:


> The system set-up i posted is by far the most transparent and powerful, i don't believe in mixing the LFE with L/R sub bass signals.


If you want to use the subs for both HT and HIFI you have no other choice. SOMEWHERE in the chain you must sum the signals. 



Mika75 said:


> Most Speakers have built in crossovers, this is what u would refer to as a 'fixed EQ'


correct.



Mika75 said:


> Your Active Mains have fixed crossovers, but also adjustable shelving filters for low/high (u should be using these)... this is all still EQ.


correct. Can be 'gain' rather than shelving filter though.



Mika75 said:


> Thinking as an Audiophile, u now have some issues.. onder:


hence the joke smilie at the end of my sentence


----------



## Wayne A. Pflughaupt

Drizt said:


> If you want to use the subs for both HT and HIFI you have no other choice. SOMEWHERE in the chain you must sum the signals.


I'd have to disagree. Stereo bass for music (if that floats your boat); LFE for movies. There's no good reason to sum them that I can see. If there was, the capability to do so would be built into every receiver.



> correct. Can be 'gain' rather than shelving filter though.


Huh?

Regards,
Wayne


----------



## sfdoddsy

Use the attenuators on the power amp to adjust overall gain levels. That's what they are there for. As a digital device, when you adjust gains using the DCX you lower the overall resolution. This may not matter when you use it just for the bass, but it does when you use for ovetrall system EQ and crossovers (as I do).

Basically, you want to feed the highest level signal you can into the DCX from your preamp. By reducing the gains on the power amp you can increased the gains from the preamp without affecting overall system volume.

BTW, I have the same preamp (well, the Onkyo version) and it does output higher levels via the XLR outputs than it does from the RCA.


----------



## Drizt

Wayne A. Pflughaupt said:


> I'd have to disagree. Stereo bass for music (if that floats your boat); LFE for movies. There's no good reason to sum them that I can see. If there was, the capability to do so would be built into every receiver.


It is in every receiver already.

You set your speakers to 'small', set a crossover point for that speaker and the LP gets sent to your LFE channel.

As for a reason on why to do so, well if the mains could do with some more bass (extension and/or volume) then sending the low frequencies to the sub(s) would help out. In an AVR this could be achieved by setting the mains to 'small' and setting the appropriate crossover point.

Now if you have this same setup when using the HT setup, then the LFE + the summed L & R signals will be sent to the sub(s). This would be mono bass.

If you want to maintain stereo bass you would need to do the 'summing' after the AVR output.


----------



## Drizt

brucek said:


> Do some reading about audio gain structure on the internet - it's one of the more important subjects in the pro equipment world.
> 
> brucek


Thank you for this advice. Knowing what to google for was very helpful.

Ive started reading up on it, and found a few good references.

http://www.rane.com/note135.html
http://www.prosonicsolutions.com/articles/Sound System Gain Structure.pdf
http://www.recordingreview.com/arti...e-Basics-Of-Setting-Gain-Structure/Page1.html
http://www.soundonsound.com/sos/apr98/articles/gainstructure.html
.. and many more when googling for 'audio gain structure'

Should be armed with enough information now to work out how to piece the system all together.

Thanks everyone.

Ill let you know how it all turns out  - (Should be in 3-4 weeks).


----------



## Wayne A. Pflughaupt

Drizt said:


> If you want to maintain stereo bass you would need to do the 'summing' after the AVR output.


 I really have doubts that this is going to work. I only have a couple of CDs in my entire collection that actually have a stereo bass signal. One of them is Pink Floyd’s _Wish You Were Here_, which has one track with bass quarter notes “pulsing” back and forth between the left and right channels to the beat. If you don’t have this, it might be a good one to acquire to help judge the success of your experiment.

Taking this track as an example: What is the result when you sum a “pulsing” bass signal that moves back and forth between the two channels? You’re going to get a “pulsing” mono signal.

Now, combine that pulsing mono signal with the original pulsing stereo signal. What are you going to get, from a purely audible perspective (i.e., what are you going to hear from your stereo subs)? A pulsing mono signal of course. Why? Because at any given moment when, for instance, the stereo feed is sending the note to the left channel, the mono feed is sending it to both the left and right channels. So you no longer have any separation. Or at best, you get a signal that “moves” only “halfway” between the two channels, not fully.

Of course I expect your intent is to blend movie bass rather than music, but if there ever is any kind of “stereo” bass signal in the movie that might appear in one channel only – let’s say the left - it’s going to end up in both subs. If not as a pure mono signal at equal levels in both subs, then for sure one that images half-way between center and left instead of full left.

Lacking a situation like that, what this set-up is going to do for the most part is merely result in additional gain, which is typically the result of summing two signals.

Regards,
Wayne


----------



## Wayne A. Pflughaupt

sfdoddsy said:


> Basically, you want to feed the highest level signal you can into the DCX from your preamp.


Is that mentioned anywhere in the DCX manual (feeding it the highest possible signal level, that is)?

Regards,
Wayne


----------



## Drizt

Wayne A. Pflughaupt said:


> Is that mentioned anywhere in the DCX manual (feeding it the highest possible signal level, that is)?
> 
> Regards,
> Wayne


Not sure if it is mentioned in the DCX manual or not, but it seems its common practice on all devices in the chain. On brucek's advice I read up on the following. It makes sense after you have read it. 



Drizt said:


> Thank you for this advice. Knowing what to google for was very helpful.
> 
> Ive started reading up on it, and found a few good references.
> 
> http://www.rane.com/note135.html
> http://www.prosonicsolutions.com/articles/Sound System Gain Structure.pdf
> http://www.recordingreview.com/arti...e-Basics-Of-Setting-Gain-Structure/Page1.html
> http://www.soundonsound.com/sos/apr98/articles/gainstructure.html
> .. and many more when googling for 'audio gain structure'
> 
> Should be armed with enough information now to work out how to piece the system all together.
> 
> Thanks everyone.
> 
> Ill let you know how it all turns out  - (Should be in 3-4 weeks).


----------



## Drizt

The only summing I intend on doing is 

1). L channel + LFE => Left sub
2). R channel + LFE => Right sub

I will not be summing the L & R channels at any stage.

So at no point will the bass that was intended for only one channel, will end up coming out of both subs.

Hope that makes more sense now for you ?

Any yes, this summing is 'only' going to happen when watching movies. During music where its 2ch only and no LFE output, then the 'summing' will not do anything.



Wayne A. Pflughaupt said:


> I really have doubts that this is going to work. I only have a couple of CDs in my entire collection that actually have a stereo bass signal. One of them is Pink Floyd’s _Wish You Were Here_, which has one track with bass quarter notes “pulsing” back and forth between the left and right channels to the beat. If you don’t have this, it might be a good one to acquire to help judge the success of your experiment.
> 
> Taking this track as an example: What is the result when you sum a “pulsing” bass signal that moves back and forth between the two channels? You’re going to get a “pulsing” mono signal.
> 
> Now, combine that pulsing mono signal with the original pulsing stereo signal. What are you going to get, from a purely audible perspective (i.e., what are you going to hear from your stereo subs)? A pulsing mono signal of course. Why? Because at any given moment when, for instance, the stereo feed is sending the note to the left channel, the mono feed is sending it to both the left and right channels. So you no longer have any separation. Or at best, you get a signal that “moves” only “halfway” between the two channels, not fully.
> 
> Of course I expect your intent is to blend movie bass rather than music, but if there ever is any kind of “stereo” bass signal in the movie that might appear in one channel only – let’s say the left - it’s going to end up in both subs. If not as a pure mono signal at equal levels in both subs, then for sure one that images half-way between center and left instead of full left.
> 
> Lacking a situation like that, what this set-up is going to do for the most part is merely result in additional gain, which is typically the result of summing two signals.
> 
> Regards,
> Wayne


----------



## Wayne A. Pflughaupt

> So at no point will the bass that was intended for only one channel, will end up coming out of both subs.


 Yes it will, if the LFE is a signal summed from the L & R and then mixed back in with them. 

But as you said, the summing is mostly for movies, so it’s essentially moot. The summing of the three signals is basically only going to result in an increase in gain to the EQ and amplifier.



> Not sure if it is mentioned in the DCX manual or not, but [feeding it the highest possible signal level] seems its common practice on all devices in the chain.


No, it isn’t. You’re supposed to allow 12-20 dB headroom in each device between the receiver and the amplifiers, and if you max out their input signals you won’t have that. You also won’t have headroom for any EQ boosting that might be used post A/D conversion.

Regards,
Wayne


----------



## Drizt

Have you had a read of the links from the above post?

Heres a snippit from one of them



> Summary
> 
> Optimum performance requires correctly setting the gain structure of sound systems. It makes the difference between excellent sounding systems and mediocre ones. The proper method begins by taking all necessary gain in the console, or preamp. All outboard units operate with unity gain, *and are set to pass the maximum system signal without clipping.* The power amplifier sensitivity controls are set for a level appropriate to pass the maximum system signal without excessive clipping. Lastly, active crossover output controls are set to correct for loudspeaker efficiency differences.





Wayne A. Pflughaupt said:


> No, it isn’t. You’re supposed to allow 12-20 dB headroom in each device between the receiver and the amplifiers, and if you max out their input signals you won’t have that. You also won’t have headroom for any EQ boosting that might be used post A/D conversion.
> 
> Regards,
> Wayne


----------



## Wayne A. Pflughaupt

I used to install pro audio sound systems for a living, so I’m familiar with gain structure. 

The section you highlighted does not say that outboard devices’ levels must be set at the highest possible level. It says the devices must be able to _pass the maximum level they are expected to receive *without clipping.*_ That’s not the same thing.

From the Rane paper linked in your earlier post:



> An examination of all audio signals reveals music as being the most dynamic (_big surprise_) with a _crest factor_ of 4-10. Crest factor is the term used to represent the ratio of the peak (crest) value to the _rms_ (_root mean square_ -- think _average_) value of a waveform. For example, a sine wave has a crest factor of 1.4 (or 3 dB), since the peak value equals 1.414 times the rms value.
> 
> Music's wide crest factor of 4-10 translates into 12-20 dB. This means that musical peaks occur 12-20 dB higher than the "average" value. This is why headroom is so important. _*You need 12-20 dB of headroom in each unit to avoid clipping.*_


Regards,
Wayne


----------



## Drizt

I think we are just using different terminology (most likely mine is wrong). Your talking about adding headroom above an 'average' and im talking about the 'maximum' value.

But..

If an abstract value of 10 is the *'highest' *value that unit A will output, then 10 should be the highest value that the unit B can accept before clipping. As it is the highest value, then theres no need to allow for any further headroom.

Are we in agreement on that? If not then im totally lost again.



Wayne A. Pflughaupt said:


> I used to install pro audio sound system for a living, so I’m familiar with gain structure.
> 
> The section you highlighted does not say that outboard devices’ levels must be set at the highest possible level. It says the devices must be able to _pass the maximum level they are expected to receive *without clipping.*_ That’s not the same thing.
> 
> From the Rane paper linked in your earlier post:
> 
> 
> 
> Regards,
> Wayne


----------



## Wayne A. Pflughaupt

Not adding headroom, _leaving_ headroom. The point is, you can’t definitively nail down exactly what “10” is. A recommended headroom of 12-20 dB – when you think about it, that’s a tremendous spread. The reason is the unpredictability factor. In a live sound situation, unless the sound engineer is dealing with a particular act on a regular basis, he has no way of knowing for sure what those guys on stage are going to throw his way. So he must leave enough headroom for a worse-case scenario.

For us at home, it’s not quite as bad. CDs for instance, they can vary in levels from one to the next, but not tremendously. And we merely adjust our volume controls for one that’s unusually loud or soft.

Movies can be unpredictable, however. The latest Batman movie _Dark Knight,_ for instance, had such severe bass levels that it practically blew out my subs. Far greater than the “reference” DVDs I had used to set up my system. The point is, you just never know what a DVD is going to dish out, so to arbitrarily pick something off the shelf as “the standard” and expect that nothing will ever come down the pike that’s more demanding is foolish.

Bottom line, modern digital pro audio gear, at least anything that’s of decent quality, has excellent dynamic range and S/N specs. You’re not going to add any background noise by allowing for some headroom, so there’s no good reason to max out the input signal levels and leave none. And again with an equalizer, headroom must be allowed for any positive-gain filters that may be used.

Regards,
Wayne


----------



## Drizt

Thank you for explaining. I must have read it wrong as the instructions I read from the links I provided detailed how to set the system up form end to end. It said to turn it up until you find the units clipping given a 'reference maximum' and then back them off a bit.

Could you please detail your steps end to end on what you would do in my situation?

Integra -> MX882 -> DCX -> Quest

Sorry to put you on the spot, but it would help me out a lot.



Wayne A. Pflughaupt said:


> Not adding headroom, _leaving_ headroom. The point is, you can’t definitively nail down exactly what “10” is. A recommended headroom of 12-20 dB – when you think about it, that’s a tremendous spread. The reason is the unpredictability factor. In a live sound situation, unless the sound engineer is dealing with a particular act on a regular basis, he has no way of knowing for sure what those guys on stage are going to throw his way. So he must leave enough headroom for a worse-case scenario.
> 
> For us at home, it’s not quite as bad. CDs for instance, they can vary in levels from one to the next, but not tremendously. And we merely adjust our volume controls for one that’s unusually loud or soft.
> 
> Movies can be unpredictable, however. The latest Batman movie _Dark Knight,_ for instance, had such severe bass levels that it practically blew out my subs. Far greater than the “reference” DVDs I had used to set up my system. The point is, you just never know what a DVD is going to dish out, so to arbitrarily pick something off the shelf as “the standard” and expect that nothing will ever come down the pike that’s more demanding is foolish.
> 
> Bottom line, modern digital pro audio gear, at least anything that’s of decent quality, has excellent dynamic range and S/N specs. You’re not going to add any background noise by allowing for some headroom, so there’s no good reason to max out the input signal levels and leave none. And again with an equalizer, headroom must be allowed for any positive-gain filters that may be used.
> 
> Regards,
> Wayne


----------



## sfdoddsy

Wayne A. Pflughaupt said:


> Is that mentioned anywhere in the DCX manual (feeding it the highest possible signal level, that is)?
> 
> Regards,
> Wayne


I'm not sure if it is in the manual or not. I suspect not since the manual is for pro use and we use the DCX and the BFD somewhat differently.

As you know, in pro use you would be expected to feed the DCX a fixed signal and do the volume control afterwards. We do the volume control before the DCX and hence feed it a variable signal that is usually quite a bit lower in level than it is expecting. When doing the A/D conversion this causes a (theoretical) loss of resolution of 1 bit for every 6dB of attenuation.

The ideal solution would be to do volume control after the DCX, but this is too much hassle for most people. So a simpler solution is to use the attenuators on the power amp to force the preamp to feed the DCX a higher signal, one that just about clips it on peaks.

Of course, how audible this is moot, especially when using the processor just for sub duties, but I use mine range.

In any case, as mentioned above you generally want you pre-amp delivering as close to maximum output as possible for the best S/N, and adjustable gains on the power amp allow this.


----------



## Drizt

sfdoddsy said:


> In any case, as mentioned above you generally want you pre-amp delivering as close to maximum output as possible for the best S/N, and adjustable gains on the power amp allow this.


Can you please detail how you would do this in practice?

Using a reference DVD seems fraught with danger to me (under estimate what the maximum would be). Apart from hooking up a signal generator, how could you simulate a maximum signal?


----------



## brucek

> Using a reference DVD seems fraught with danger to me


What danger? 

We've already established there is no danger with the amp safeguards in place. Set your levels as I explained earlier. 

This is not a fixed level system (as pointed out by Steve) - it is subject to the vagaries of a volume control and your personal listening habits. That means the absolute maximum peak that the system will be subject to is a movie with a lot of bass LFE. 

Play the movie at the craziest level you'll ever use - watch the meters that they are just at or below clipping. That's it. Yes, regular listening will not make use of every bit through the ADC/DAC chain - it's not that important in the sub world, and there is absolutely nothing you can do about. You have to adjust your gain structure to accommodate a volume control at the source.

brucek


----------



## spearmint

Hello Drizt,

I've just had a quick read of this thread, and IMO you're making a mountain out of a mole hill.

If you are using a balanced connection from your pre to the rest of your components then IMO you have nothing to worry about. There has been some fantastic advice given in this thread and I feel you should go back and re-read it. 

The only device in your chain that is reputed to have level issues (that I'm aware of) is the DCX2496 where some have intimated the ADC's can induce noise into the signal if fed consumer levels, how much of this would be an issue with subs I've no idea. Having the MX882 in circuit before the DCX2496 will allow you to feed the DCX correct levels, after that into a pro amp shouldn't be an issue.

Good luck with it all, and don't be afraid to experiment.


----------



## Drizt

spearmint said:


> I've just had a quick read of this thread, and IMO you're making a mountain out of a mole hill.


You know me well. When do I not over react / over think things ? 



spearmint said:


> Good luck with it all, and don't be afraid to experiment.


Im always afraid of making mistakes. Its one of my biggest failings 

Ill give it a go and see how it all goes.


----------



## Wayne A. Pflughaupt

Drizt said:


> ...the instructions I read from the links I provided detailed how to set the system up form end to end. It said to turn it up until you find the units clipping given a 'reference maximum' and then back them off a bit.


I haven’t read through all the articles, but I expect there they were mainly referring to the system front-end and back-end – i.e., the mixing console and the amplifiers.

I started out to write a gain structure primer for our Forum a few years ago, but ultimately gave up on the idea. After studying that Rane article and others, it struck me that most of what they said about how proper gain structure should be done simply doesn’t cross-reference well to a home audio system. 

For example, the Rane article notes that in live audio situations they’re primarily concerned about headroom, not dynamic range, since the latter is mainly determined by ambient noise levels in the venue and as such beyond their control. By contrast, inadequate headroom that results in clipping and distortion is the “kiss of death,” virtually assuring they will lose a paying client.

So you can see that their priorities are virtually backwards compared to ours in home audio. For us dynamic range is rather critical, while headroom is a minor issue, due to a relative uniformity in the equipment we use (more on that shortly).

Most info on the gain structure subject notes how important it is to set the proper level for each input to the mixing console. Their situation is that everything plugged into the console has a different signal level – keyboards vs. mics, etc. Some mics deliver hotter signals than others; the variance from one make and model to another can be pretty extreme. A guitar or bass guitar with active (battery powered) pick ups will have a hotter signal than one with passive pick ups. And on and on. 

The idea is to maximize each source’s signal-to-noise right up front, using the gain control that each input of the mixing console has. This allows them to bring some uniformity to these various mismatched incoming signals, so that they all have a uniform level at the console’s outputs. 

Once again, this is not relevant to domestic audio systems at all. With consumer equipment, there is a fairly universal standard for signal levels from top to bottom - from the source media, to the CD, DVD players, DVRs, etc. that play back the media, and on down the line. This is why you very seldom see input gain controls on home audio pre-amps or amplifiers: It is not expected that they will ever see a wide variation of input signal levels. It’s taken for granted that they will never see a +26 dBu input signal from a professional mixing console, for example, or a few-millivolt signal from a microphone.

So what do we get when we insert a piece of pro audio gear into our “universal standard” signal chain? Well, we have a bit of a problem going in, because unlike the mixing console, our front-end signal levels are pretty much set in stone. They’ll never exceed a certain level, and that level is considerably lower than what pro gear is designed for.

And there is another issue. Most home gear has A-weighted background noise specs (i.e. S/N ratio), which is a less rigorous standard than the unweighted specs most good-quality pro gear adheres to. The problem with A-weighting is that it “ignores” lower-frequency noise that might be present. So if a receiver or DVD player happens to generate a bit more hum than it should, and this problematic component can only muster an unweighted noise spec of 88 dB, applying A-weighting could allow the manufacturer to “honestly” bump the figure up to a more respectable 93 dB, for example.

So – let’s drop a pro audio processor into our A-weighted, low-level signal chain. What will it get you to boost the signal level up? Sure, the oft-touted wisdom says it's supposed to maximize the dynamic range of the pro processor. But you’re also boosting any noise levels that might be present in the signal, so have you really made any improvement?

Fortunately our home gear has something of an “ace in the hole,” but with caveats. The reason products like the BFD have +4 dBu/-10 dBV level switching is that there is a 20-odd dB difference in nominal operating levels between pro and consumer gear, so one internal gain structure cannot be optimal for both. In other words it’s impossible to deliver, at the same time, the best headroom and the best noise floor from a single gain structure, hence the switchable ranges: -10 dBV offers the lowest noise floor, while +4 dBu gives the most headroom. Fortunately, these days good-quality pro gear is quiet enough that manufacturers are able to largely ignore this lack of optimization and stick with the +4 dBu structure.

And here’s our “ace in the hole:” All things considered and A-weighting aside, you’re going to find that good-quality home receivers and pre-pros have a quieter background noise level than good-quality pro gear that operates only with the +4 dBu gain structure. There’s not a huge difference between them, but it does exist. That certainly gives us some leeway when jacking up the signal we’re feeding to a pro processor. 

But here’s the caveat: Boost up the output signal level of the home pre-pro or receiver, and at some point the noise floor inherent in that signal will exceed the pro processor’s innate noise floor. Naturally, when that happens it’s going to be passed along to everything downstream in the signal chain.

Also consider: If you introduce a pro audio processor to your signal chain and it has a higher noise floor than the rest of your system, then its noise floor now determines that of your entire system. It can never be lower than the weakest link in the chain, and boosting the incoming signal from a component that’s inherently quieter is not going to change that. 

And here’s another caveat: The receiver or pre-pro may only be quieter than the pro processor when in a straight or bypass mode. Engage any digital processing like Dolby Digital, and the difference between the two will quite possibly disappear, as digital processing adds noise (naturally, how much is added will depend on your particular receiver or pre-pro). Of course, this only increases the possibility that kicking up the signal level too much will increase overall system noise levels beyond that of what the pro audio processor contributes.

Are you starting to see why I ditched the idea of a gain structure primer for mixing and matching home and pro gear? The whole thing is a convoluted mess.

But where does this leave us? Fortunately as we shall see, things aren’t as bad as they appear. 

With all this in mind, let’s refer back to the Rane paper:


> Many outboard units operate at "unity gain," and do not have _any_ level controls -- what comes in (magnitude-wise) is what comes out. For a perfect system, _all_ outboard gear would operate in a unity gain fashion. _It is the main console's_ (or preamp's) job to add whatever gain is required to all input signals. After that, all outboard compressors, limiters, equalizers, enhancers, effects, or what-have-you *need not provide gain beyond that [which may be] required to offset any amplification or attenuation they may provide [as a result of their specific function].*
> 
> All level controls in outboard units (except active crossovers: see below) exist primarily for two reasons:
> 
> They provide the flexibility to operate with all signal sizes. If the input signal is too small, a gain control brings it up to the desired average level, and if the signal is too large, an attenuator reduces it back to the desired average.
> Level controls for equalizers: the need to provide make-up gain in the case where significant cutting of the signal makes it too small, or the opposite case, where a lot of boosting makes the overall signal too large, requiring attenuation.
> 
> Whether the system contains one piece of outboard gear, or a dozen, gains are all set the same way. Again, the rule is to maximize the S/N through each piece of equipment, thereby maximizing the S/N of the whole system. And that means setting things such that your _maximum system signal_ goes straight through every box without clipping.


In other words, as long as the outboard processing gear can pass the maximum-expected signal straight through without clipping, then *maximum system S/N has been achieved.* And basically, _any input signal metering that’s both “on the charts” and below clipping fits the bill._ 

At the end of the day, it’s as simple as that. The only reason to be concerned about signal levels with outboard processors is if they are exceedingly low (like they never register) or so high that they cause clipping. That’s it. 

And in the case of the former, I’m not even sure it’s a viable issue for us, since low signal levels are part and parcel to home audio gear. Feed a -10 dBv signal to a +4 dBu processor and you can expect low meter readings. That’s just the way it is. But boosting it up doesn’t necessarily mean you’re going to improve dynamic range or S/N. In fact as noted, you may make it worse.



> Could you please detail your steps end to end on what you would do in my situation?
> 
> Integra -> MX882 -> DCX -> Quest


Considering your particular situation with the MX882 and DCX, and how to set their respective levels - using the MX882 like you intend makes it something of a queer duck in the signal chain. Basically, the MX882 is a stripped-down mixer, compensating for a wide variety of input signal levels without all the AUX sends, EQ capabilities, etc. that you find on a full-featured console. The problem is, in your situation it’s being inserted in the signal chain _after_ the pre-amp, rather than at the front of the signal chain like a mixer normally would be. As noted, boosting the incoming signal from the pre-pro or receiver is viable only to the point where any residual noise it carries begins to swamp the MX882’s noise floor. Fortunately in this case that’s not a big deal, since we’re dealing with subwoofer signals.

I think what I’d do is just set the MX882 gain controls for unity (straight-up zero) and expect that you may get low readings on its meters. If you’re not comfortable with the levels the downstream DCX is receiving, then use the MX882 to kick them up. Basically if the meters for either piece are somewhere in the green at normal listening levels, you’re fine on all counts: S/N and headroom, you have enough of both.

As far as the amp goes, the standard MO for a home audio subwoofer amplifier is to adjust its gain settings until your subwoofers are blending with the rest of the system at the level you desire. If perchance the amp’s inputs clip at this gain setting, then back down on the MX882 until there’s no more clipping and re-adjust the amp’s gains. Or, if the sub levels are inadequate with the amp gains all the way up, then dial up the MX882’s levels.

Regards,
Wayne


----------



## Drizt

wow Wayne, that is an awesome post.

Thank you very much for taking the time to write that.

Can I ask one last question?

From reading between the lines in posts here, perhaps Im using the wrong items for the job? If I scrap the idea of stereo bass (From the subs for 2ch) and just use the subs for the mono LFE channel on HT usage, then would I be better off just getting the BFD ?


----------



## spearmint

Wayne thanks for the fantastic post. :T



Drizt said:


> From reading between the lines in posts here, perhaps Im using the wrong items for the job? If I scrap the idea of stereo bass (From the subs for 2ch) and just use the subs for the mono LFE channel on HT usage, then would I be better off just getting the BFD ?


Drizt, the FBQ is great for EQ; however you cannot use it to set a shelving filter if you're requiring an LT.


----------



## Drizt

spearmint said:


> Drizt, the FBQ is great for EQ; however you cannot use it to set a shelving filter if you're requiring an LT.


Thanks Spearmint. It all gets a bit confusing rather quickly for the noobie's like me  I spoke to A9X and he suggested I probably could get away without an LT with my subs and just EQ the response a bit to tailor it to what I want. Originally from my reading I thought an LT would be mandatory, but perhaps its just my level of understanding is not up to scratch.

I do apologise if people are getting frustrated at me for what seems like being slow to pick this all up.


----------



## Wayne A. Pflughaupt

Drizt said:


> wow Wayne, that is an awesome post.
> 
> Thank you very much for taking the time to write that.
> 
> From reading between the lines in posts here, perhaps Im using the wrong items for the job? If I scrap the idea of stereo bass (From the subs for 2ch) and just use the subs for the mono LFE channel on HT usage, then would I be better off just getting the BFD?


Thanks for the kind words. This topic has been a thorn in my side for a few years now and has needed to be addressed. Fortunately today everything that's been floating around in my brain finally fell into place into something hopefully reasonably coherent. I may tackle that primer after all. 

Regarding the BFD - sure, it’s the most logical choice if all you need is basic sub equalization. If that’s the case, then the DCX’s capabilities are going to waste. But I don’t want to discourage you from experimenting. It’s always fun and usually educational as well. Besides, I’d like to see if all my theories as to why the L-R/LFE combining scheme won’t work will actually hold water. :laugh:



Drizt said:


> I do apologise if people are getting frustrated at me for what seems like being slow to pick this all up.


No problem. What, you think we were born with all this stuff in our brains? We had to learn just like everyone else. :T

Regards,
Wayne


----------



## Drizt

Wayne your an absolute champ 

Ill give it a go. As brucek says, I have nothing to worry about.

oh.. one last question (i bloody hope).

Can I hook everything up EXCEPT for the subs to test all the levels? Eg. Can the power amp be turned on but not hooked up to the subs to see if the inputs clip?

I only ask as I used to own some tube amps that 'had' to be hooked up to a load or you would damage.


----------



## Wayne A. Pflughaupt

Sure, it's no problem disconnecting the speakers when calibrating your signal levels. In fact, I believe the Rane paper recommends this, to avoid damaging the speakers or people's hearing.

But it doesn't matter as much that the amp inputs clip as whether or not you have all the volume you need from the subs before that happens. The procedure I mentioned in my last post will work fine for setting the amp gains.

BTW - spearmint, forgot to acknowledge you in my last post, but thanks for the kind words. :T

Regards,
Wayne


----------



## Mika75

I'm still a little confused Dritz, why did u decide on a sealed design that's so limited in extension ?

Seeing as ur targeting 'Movie' sub bass and ur mains are more than capable down to 20Hz, An LLT utilising one Mal-X would have been much better suited to ur needs. 

My quick calculations show a Sealed Mal-X with an LT circuit will max out around 110db @ 20Hz *(falling even less below that), whereas an LLT will be coasting along producing 118db @ 20Hz, and approx 115db @ 13Hz 

*calc doesn't take into account room boost / and please let me know if my measurements are way out


----------



## Drizt

Why, because I wanted a non intrusive sub (x2) as I'm not interest in massive subs for a shared lounge room. That said they are still going to be pretty big and to some people (including my wife) they will be too intrusive in appearance. As for output, 110dB at 20Hz each sub is plenty for me. Not sure why anyone would need more to be honest. But I understand that more headroom is always best. 

I am not just targeting HT bass. Im also going to tinker with them for 2ch use. I wanted to keep my options open for stereo bass, so this necessitates having two subs.




Not everyone wants massive ugly subs. Those that do good luck to them. Probably a bit harsh, as they wouldn't think of them as ugly.


Above all else I just wanted to do it. And thats enough of a reason for me. Sure theres always something better out there. But why can't we just be happy with people doing something different?


----------

