Jump to content

Gain matching amps on Biamp system


Recommended Posts

2 hours ago, Khornukopia said:

After stepping way back, I am curious about what these "gain" specs reference.

Gain


Normaly expressed in decibels, dB 
Power Gain (dB) = 10 Log 10 (RF Output Power / RF Input Power)

Gain is defined as the ratio of the output power to the input power in dB. Assume that the input power is 10 mW (+10 dBm) and the output power is 1 W (1000 mW, +30 dBm). The ratio will be 1000/10 = 100, and the gain will be 10 * log 100 = 20 dB.

Link to comment
Share on other sites

On ‎7‎/‎25‎/‎2019 at 12:47 AM, Jim Gregory said:

I use a First Watt SIT3  “11.5 dB gain” for the HF (Klipsch K402 w 3” Radian 760 drivers) and a pair of Pass Labs Aleph 2 “20 dB gain” for the LF (Klipsch 904 w k45k drivers)

… 

Any suggestions or other reasonable alternatives to gain balance the amps is greatly appreciated. 

 

16 hours ago, Jim Gregory said:

Gain


Normaly expressed in decibels, dB 
Power Gain (dB) = 10 Log 10 (RF Output Power / RF Input Power)

Gain is defined as the ratio of the output power to the input power in dB. Assume that the input power is 10 mW (+10 dBm) and the output power is 1 W (1000 mW, +30 dBm). The ratio will be 1000/10 = 100, and the gain will be 10 * log 100 = 20 dB.

 

 

So the numbers may refer to each amplifier's total output, not the difference of input sensitivity? 

 

Back to your original request, as suggested earlier, remove the in-line attenuators and adjust the level of your bass amp and tweeter amp using the front panel controls on the Marchand.

Link to comment
Share on other sites

8 hours ago, Khornukopia said:

So the numbers may refer to each amplifier's total output, not the difference of input sensitivity? 

 

Back to your original request, as suggested earlier, remove the in-line attenuators and adjust the level of your bass amp and tweeter amp using the front panel controls on the Marchand.

Thank You for taking the time to help me out with my biamp situation. I am fairly new with all this and at times have a difficult time explaining myself clearly with the proper audio terminology 

Which is why I am hesitant to ask questions in this forum because you guys are really Smart and I have to do research to answer questions y’all ask me. Lol 

I did as you suggested, removing the RCA attenuators and adjusting the HF/LF on the Marchand going to each amplifier. It sounds wonderful !!!

Auralic Aries will have Eq capabilities through the Lightning DS app. I will still need to use REW to take measurements. That will be my next step. 

Again greatly appreciate your help. 

  • Like 1
Link to comment
Share on other sites

9 hours ago, Khornukopia said:
On 7/25/2019 at 6:54 PM, Jim Gregory said:

Gain



Normaly expressed in decibels, dB 
Power Gain (dB) = 10 Log 10 (RF Output Power / RF Input Power)

Gain is defined as the ratio of the output power to the input power in dB. Assume that the input power is 10 mW (+10 dBm) and the output power is 1 W (1000 mW, +30 dBm). The ratio will be 1000/10 = 100, and the gain will be 10 * log 100 = 20 dB.

 

 

So the numbers may refer to each amplifier's total output, not the difference of input sensitivity? 

 

It's just the voltage gain ratio from input to output, with the output devices sized to handle the power being dissipated with varying loads.

 

If the hf is 10 dB hot from the lf at the same drive levels on the same amp, and another amp is available with 10 dB less gain, it'll match up perfectly in that respect as the companion amp for the hf along with the original (higher gain) amp for the lf.  This, of course, not taking level adjustment capability of the crossover unit into consideration.  At least you can allow both adjusters to remain in a more stable position when your levels are better matched between amps.

Link to comment
Share on other sites

56 minutes ago, Jim Gregory said:

I did as you suggested, removing the RCA attenuators and adjusting the HF/LF on the Marchand going to each amplifier. It sounds wonderful !!!

Auralic Aries will have Eq capabilities through the Lightning DS app. I will still need to use REW to take measurements. That will be my next step. 

 

I haven't looked it up.  Will this be a digital or analog equalization?  And I guess I don't really care (rather, have an opinion) either way, but if you're entering the digital domain, why not do the crossovers there, too, and in fact fell both in one swoop?  It'd be fun to play with finite impulse response filters to better tame the phase shifting

Link to comment
Share on other sites

2 hours ago, glens said:

I haven't looked it up.  Will this be a digital or analog equalization?  And I guess I don't really care (rather, have an opinion) either way, but if you're entering the digital domain, why not do the crossovers there, too, and in fact fell both in one swoop?  It'd be fun to play with finite impulse response filters to better tame the phase shifting

I am fairly confident that the equalization being adjusted through Lightning DS is digital then converted through the Auralic Vega G2 to Analog. I have quite a bit invested into the Marchand acoustical electronic crossover and right now happy with the results. 

Sorry, I do not have any knowledge on phase shifting so like many other aspects of this hobby I will have to find literature on the subject and attempt to educate myself. I do find these audio technicalities you guys discuss fascinating. I wish I knew where I could take a Audio course online that could guide me through this discipline. Thanks to kind folks like yourself I get help for my system and insight on new information to explore. 

Thank You for your help Sir. 

Link to comment
Share on other sites

Basically:

 

For every element in a filter, there is a 45 degree change in separation between the voltage and current waveforms passing through the circuit.  (They start out "in step" with each other and ideally would remain that way.)  In a passive filter, which is not what you'll be using but it's easier to explain directly - they function the same as what you'll have.  A capacitor in series with the driver will, based on the values of the capacitor and the driver cause the response going to the the driver to decrease as frequency goes lower.  (The capacitor gets higher "resistance" as the frequency drops - but it's not "pure resistance" and that's the "problem.")

 

A single element like this will attenuate frequencies at a rate of 6 dB per octave and will allow the current to get through to the driver in the normal time relationship, but will cause the voltage to delay by 45 degrees.  Neither the current nor the voltage will singly do anything - they work together.

 

Adding an appropriate inductor in parallel between that capacitor and the driver will cause a further 6 dB per octave attenuation, and a further separation of 45 degrees between the current and voltage waveforms.  Put another capacitor in series after, 6 dB steeper-yet filter and another 45 degrees separation.  Yet another coil in parallel after that, another 6 dB steeper filter with yet further phase separation between current and voltage by 45 degrees.  At this point we've got a 4th-order high-pass filter (24 dB / octave attenuation) and the current leading the voltage by 180 degrees (out of phase with each other).   

 

The complement to this filter is for the lower-frequency driver and has the opposite effect in every respect.  The current lags the voltage by 45 degrees per element.

 

So now, with 24 dB / octave filters, which is what the Linkwitz-Riley filters are in your unit, the current waveforms of the high and low pass portions are 360 degrees out of phase with each other, which is to say they're back in phase, but at one cycle apart.  The current waveform is what's doing the work to cause sound, but current without the voltage to carry it does any good - it remains merely potential.

 

With complementary 2nd-order filters (12 dB / octave), the two current waveforms are 180 degrees out of phase with each other, thus the driver outputs are out of phase with each other at the crossover point and it's common to reverse the polarity of one of the drivers so they don't try to cancel each other out within their overlap.

 

Your unit is performing the filtering at a low voltage level, not using inductors, but the response and phase characteristics are the same.  And I'm a fan of Linkwitz-Riley alignment, so as far as that goes I think you're golden.  But in other respects your unit is outdated.  You need a crossover and it does that admirably enough, but you also need shaping, which it doesn't do.  This can certainly all be done in analog, but it doesn't make best sense if digital information is also being handled.

 

In a digital Infinite Impulse Response filter, the phasing changes in the same way as described above.  Finite Impulse Response filters require more factors, computation (processing horsepower) and time, but they trade in part for phase stability.  Some say the phase changes aren't objectionable, but I'd like to (some day) make that determination for myself.  And digital processing power is as cheap now as is solid-state amplification.

 

So you'll be doing at least some response shaping to flatten (maybe better to say "untilt") the response of a constant-directivity horn, but this will only be available to you for digital sources, and possibly if not likely rather incomplete anyway.  Do you have analog sources, and if so, what do you plan to do to accommodate them in this respect?

 

You could gather the parts and implement the response shaping (bleed off excess energy) with passive components in parallel with the hf driver.  But that could be just as expensive as getting a digital crossover which can also effect the in-band shaping.  And the passive parts wouldn't do as complete a job as easily as the digital gear.

 

I commend you for what you want to get done, I just don't think it's cost- or result-effective enough for the money.  But that's for me, not you.  Just trying to help you see the bigger picture.

 

Link to comment
Share on other sites

33 minutes ago, glens said:

Basically

Glens

Thank You for taking the time to break down what to me is very technical information. I do understand some of what you are explaining but I will have to look up some of the terminology to really grasp the concept then put it all back together in order to have a cohesive understanding. I do think I get the bottom line, “if I don’t do something to “response shape” the high frequency signals that are being sent to the constant directivity horn “ the Klipsch 402” then I can not achieve the optimal potential sound quality the 402’s have to offer. 

I do not use any analog sources and do not plan to do so in the future. I only stream digitally. I only recently acquired a basic understanding of what Equalizing Frequencies means and  considered measuring and application of Eq. And now having read your explanation above  realize the necessity. I really believe digital software is the best path. But this was my problem and This may not be correct, but I was reluctant to put a digital Eq device “that converted the signal to analog” in front of my Vega G2 which is a much better DAC. I looked into an analog Eq component “Charter Oak “ but seemed to me to be more of a subjective approach. To further complicate matters I only use my iPad Pro with the Auralic Lightning DS app with Tidal as my music source. I do not mind spending some $ to accomplish the response shaping but I will need a lot of expert advice on how to best achieve a positive outcome. I really do appreciate you looking at my situation and your willingness the help a total stranger. You are obviously very educated in this field, however if my situation is more than you bargained for I completely understand. At this point I would love to fix the problems that I don’t even possess the knowledge to understand “lol”. Apologies for the lengthy explanations and I will continue to revisit your explanation on “phase shifting” till I can converse somewhat intelligently on the subject. 

Kind regards. 

Link to comment
Share on other sites

So you want to make use of the DAC you have and like/prefer instead of one in, say, a MiniDSP HD.  That's a quandary and certainly imposes limitations.

 

Any chance you can develop a complex digital shaping-filter which your DAC can apply to everything it handles?  I'm unfamiliar with that unit and its capabilities, other than what I've gleaned from what you've said.  A more simple software equalizer could get fairly close, but if your DAC (and I assume streamer?) is running a form of GNU/Linux (as likely as not) then it may be possible to route everything through a specific filtering arrangement prior to producing the analog output to your crossover.  Or maybe your unit has output and input for a digital processing loop?  At least you'd be able to get the overall frequency response right where it needs to be and let all the other chips fall where they may.  They're generally considered to be smaller, less important "chips" anyway.

Link to comment
Share on other sites

19 minutes ago, glens said:

ny chance you can develop a complex digital shaping-filter which your DAC can apply to everything it handles?

One of the developers of the Auralic products is Xuanqian. He frequently answers technical questions in regards to Auralic products. You are correct that the Auralic Vega G2 is a  streamer/dac and preamplifier. The Vega has no processing engine inside of it but it’s Brother the  Auralic Aries G2 does. I am going to compose a message to Xuanqian in the Auralic community forum using the information / questions you have given me and will get back to you on his response. 

Thank you again Sir for your willingness to me. 

Link to comment
Share on other sites

  • Moderators

I had a problem with gain from consumer to pro, from source to the active Crossover for a biamped system.

 

Ended up fixing it with something someone here suggested, it’s called Art Clea Box Pro, it’s made to make the gain adjustable and worked great with no noise. I use it with th 402 horns, so if there were any noise I would hear it.

 

When Klipsch was having problems trying to gain match a tube amp an a ss amp in there listening room Roy asked me the name of that unit since he seen it work. 

They ordered one an it worked for them, anyone listening in there listening heard it, no added noise.

 

I’m posting from an I pad or I would put a link, it was like $49 and is designed just for gain matching.

Link to comment
Share on other sites

54 minutes ago, dtel said:

Art Clea Box Pro,

Thank You dtel 

I will look it up now. It seems I have a even bigger dilemma with figuring out a way to “response shape” the high frequency signals that are being sent to the constant directivity horn “ the Klipsch 402”.  Glens has been very kind in trying to help me come up with a workable solution using my existing dac and crossover network. 

Link to comment
Share on other sites

1 hour ago, dtel said:

Art Clea Box Pro

Dtel thank you for the Art Cleanbox Pro tip. I looked it up and that was originally the type of gain adjuster component I was hoping for. Nelson Pass under First Watt made a in-line component for adjusting the gain on mismatched amplifiers called the B2. Mark at RenoHifi said the B2 would solve my problem but First Watt didn’t make them anymore and I have been unsuccessful finding one for sale in the secondary market. I have gain matched using my Marchand Level adjustments but to do so I had to both boost the HF and attenuate the Mid High (LF) Level. 

Link to comment
Share on other sites

  • Moderators

Was not sure if it would help you or not, I was surprised to find it. It is not that often in this hobby you find just what you need and at a good price.

 

but if you consider it be sure to read about the connections, it can be used to either increase or reduce the gain, make sure the connections match which way your going, or just use adapters?

Link to comment
Share on other sites

1 hour ago, dtel said:

it can be used to either increase or reduce the gain, make sure the connections match which way your going, or just use adapters?

Hello dtel

I read the Art CleanBox user manual. 

“Loop balanced outputs to balanced inputs to provide an inline unbalanced level / gain control.”

I will use xlr balanced output connections from my Marchand Mid High (LF) Into the Art  xlr balanced input connections then back out of the Art  xlr balanced output connectors to the xlr balanced inputs of my amplifier. 

Does that sound correct ?

Link to comment
Share on other sites

12 hours ago, Jim Gregory said:

"Loop balanced outputs to balanced inputs to provide an inline unbalanced level / gain control.”

I will use xlr balanced output connections from my Marchand Mid High (LF) Into the Art  xlr balanced input connections then back out of the Art  xlr balanced output connectors to the xlr balanced inputs of my amplifier. 

 

What I got out of it would be that to use it as a balanced-in-to-balanced-out level control you'd have to do the opposite of what you quoted.  Balanced out of the crossover to the Art, then unbalanced out of that side to unbalanced in to the other, from there balanced out from there to the amp.  The unit is primarily a balun which has the capability to work both ways which makes it suitable for inline level matching for either form by performing whichever conversion and back again.  Doing so will double up on any noise/distortion introduction by the unit.

 

https://www.amazon.com/Balanced-Unbalanced-Line-Level-Converter/dp/B00NK7OVA0 appears to do the same for less $.

Link to comment
Share on other sites

On 7/27/2019 at 2:01 PM, glens said:

At least you'd be able to get the overall frequency response right where it needs to be and let all the other chips fall where they may.  They're generally considered to be smaller, less important "chips" anyway.

Here is what I found out from Auralic Aries forum:

I have an Aries running the 6.0 firmware and it does indeed have a 20 band parametric equaliser. This includes peak/dip, high pass, low pass, band pass and band stop functions. There’s also a speaker placement compensation function, which allows you to compensate for situations where the left and right speakers are not equidistant to your listening position.

Use REW to measure the room acoustic then apply change. 

The equalizer lays out a graph where you can can depict the y axis range as +/- 12db or +/- 24db.

You then add points along the x axis, at each of the frequencies you want to adjust up or down. Each point you put down is placed in the middle of the frequency band that you want to adjust. You then specify the width of the band you want to adjust (the so called Q factor) and then specify the gain, + or -, in terms of dB’s.

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

×
×
  • Create New...