Setup In Home Audio Application

Dotsystem

I am using a 260 to provide bandpass filtering and PEQ to my woofers in a biamped 2 channel home audio system. My midrange has no crossover and is not handled by the 260.

I have read about setting the gain and the procedure described seems to rely on visual indicators to set the gain and limiters. My application does not use a mixer ahead of the 260 and have no clipping indicator for the amp. I have no visual feedback of output level on my cd player which is driving the 260. I do know it is presenting around 3.5 volts at full output. The bass amp input impedance is about 8Kohm and voltage gain is 27 db. It is rated at 500 watts into 4 ohms.

I need to apply some gain to blend properly with the midrange and am wondering if I have done it in the best way. I have set the bass input level for each channel in each input mixer to 0 for that input and -INF for the other input and the master level for each mixer to 0. I apply about 14 db of gain in the filter bandpass module.

Can anybody recommend an approach for how to best set the gain and limiters for least distortion and best S/N ratio in a non-proaudio configuration? Should the gain be set only in the input mixer or filter module or distributed across both?

Thanks

Bob

Dra

CD player directly into the 260? No pre-amp? Someone that might be a good source for this info might be Kevin Pippen. He hasn't been around in a while.

Gadget, heard from K lately?

3.5 V at full output on the CD? Can you control the output on it? Are you going to control the listening volume with the 260?

DRA

Dotsystem

DRA

The CD player has a hybrid attnentuator built in. The first 40 db of attentuation are performed in the analog domain. The player has 2 sets of balanced outputs. So the overall configuration is working. I just cannot be sure that I have it set up to provide the best S/N ration and lowest distortion since the recommended pro-audio set up guidelines cannot be followed.

I did find the on-line limiter calculator - no longer at the original FAQ link but available here:

http://www.poulpetersen.dk/Appn/gblimthc.html

I think I can validate the results by turning the amp off, cranking up the front end and observing the 260's output and threshold meters and make further limiter adjustments if needed.

My remaining questions are:

1) Should all the gain I need to blend the drivers be applied in the input mixer or in the filter module or some gain in both (how much in each) or does it matter at all? The goal is best S/N and least distortion?

2) I have not moved the internal jumper so it should be set to medium gain. How would I know if this should be changed for best S/N and least distortion?

Thanks for any help.

Bob

Dra

1) Make a CD with a track of silence and use your ear to track noise. If you have the RTA mic you could use the RTA function to actually measure the noise (put the mic right at the speaker). Try -0-, -0-; -0-, +14; +14, -0-; +7, +7; etc The input mixer is design to bring up the entire signal and the x-over gais are for the band output, but since you over use one output they do essentially the same thing, but at different stages.

2) Not sure. Adding gain is adding gain, whether by a jumper or by a fader, but if you are getting 3+volts that is what the 260 expects. I wouldn't change the jumper.

Are you using the GUI?

DRA

Dotsystem

DRA

Thanks for the suggestions.

I will try to find a silent track. If I create one, I have the problem of introducing noise from the soundcard, PC, etc.

In configuring the input mixer I will not blend the inputs so the opposite input fader will always be set to -INF. Once I make the CD I will experiment with distributing the gain bewteen the input mixer and the filter modules. Since the limiters appear after these modules in the block diagram, the limiters should not be effected by the gain distribution.

I am using DriveWare.

Bob

Dra

\"Pausing\" can work well also for a silence test, unless your player mutes the outputs, but I don't know why it would do that.

DRA

Dotsystem

I find that Audacity will allow me to create silent files. Now I need some quiet time to experiment.

Another thing to consider is the adjustment granularity for gain. It appears that setting gain via the filter module allows 1/10 db adjustments over some of its range. This can be helpful for blending the speaker output versus being limited to 1 db adjustments. So that could be a reason to set some of the gain in the input mixer while staying in the 1/10 db adjustment range in the filter module.

Bob

Gadget

Hi,
Been really busy getting the system ready for this weekend... One thing I have a problem with here is how are you dealing with the delay through the 260, if your running the mids outside of the 260?

If you have access to an oscilloscope it's pretty easy to find the clip point of any given amp...
G

Dotsystem

Gadget wrote:

Hi,
Been really busy getting the system ready for this weekend... One thing I have a problem with here is how are you dealing with the delay through the 260, if your running the mids outside of the 260?

If you have access to an oscilloscope it's pretty easy to find the clip point of any given amp...
G

Gadget

No scope so the the online limiter threshold calculator will have to do. I also have found a calculator to compute phase degrees based on delay.

Any idea what the propagation delay for the 260 is?

Thanks

Bob

Gadget

Only that the LR24 top to sub is just under 7.5 ms...the tops need to be delayed... most processors have around 5-8ms of delay... the more expensive the processor... the less delay... :?: :?: :?: but that is an observation, not a fact...Why are you not considering processing the whole signal? I mean I too am a purist.. of sorts....I have analog to the enth degree... but processing and crossing a signal to different components requires a precision that analog (or in your case NO xover for the tops???) crossovers could never approach.. even the REALLY hi-precision ones...Beware the driver left to reproduce all it can... for the xover region has many pitfalls which can degrade the sound in the overlap region...I would like to hear your methodology, or at least your theory here...

Gadget

Dotsystem

Gadget

I have nasty room problems around 40hz and need PEQ. It is also useful to have adjustable gain when using dissimilar amps.

I should clarify that there is a single capacitor serving as a crossover between the midrange and tweeter. There is no high pass filter to limit the low frequency response of the midrange driver.

The factory default speaker configuration uses a a Hypex plate amp employing a 4th order BW high pass and 3rd order LR low pass. This must be introducing some delay. The manufacturer claims a full bandwidth group delay < 4 ms for my speakers and < 5ms for their flagship model which also uses a plate amp.

For customers wishing to bypass the plate amp, the manufacturer recommends the 260 or the Rane PEQ55. I chose the 260 for its ease of use (Driveware). The Ashley Protea series and an FDS model appear capable as well.

My CD player's transport and DAC will handle up to 192/24 and I have a bit of 48, 96 and 192 program material. I am reluctant to use lower rated DACs for the midrange/treble but am tolerant for the < 60hz range where the room mode is a bigger problem than a slight loss of resolution. My ears are much more sensitive at higher frequencies and I would need a unit with better DACs. I know of people that have taken Behringer units and had them rebuilt so they would sound acceptable managing all drivers for other speaker systems. Not a big fan of Behringer though.

Assuming delays from the speaker are perfect, the home venue room introduces its own can of worms.

I am using 3rd order bessel bandpasses. Does this present a better or worse delay scenario? I can probably eliminate the highpass filter. Do you know the minimum delay for a 260 with no filter engaged?

Thanks

Bob

Dennis

Rational Acoustics is in Putnam CT. How far is that from your location? Instead of all the guess work on delay, have a pro come and measure it. For the price of a phone call, you could get a quote from them. Their number is 860-928-7828. If they are too far away, maybe they know of someone closer to your area.

Dennis

Dotsystem

Dennis

Putnam is a haul and I am sure there are closer alternatives but thanks for the suggestion. I have some room measuring tools. I am uncertain that better measuring tools are required. The advantage of perfect time alignment can be be overshadowed by room mode problems that will exist without PEQ.

The audibility threshold of delay applied to bass frequencies is subject to speculation. In this discussion, it is suggested that at 50hz the threshold may be as high 9.5 ms - greater than the 7.5 ms cited above.

http://www.trueaudio.com/post_010.htm

Bob

Dennis

You said \"The advantage of perfect time alignment can be be overshadowed by room mode problems that will exist without PEQ. \"

Why can't you have both? Why can't you put your mids/highs through a band in the 260 but add no filters so they are still wide open like they are now? You would then have the ability to delay the mid/hi if you were so inclined. More importantly, you can time align at the crossover point and at least partially eliminate the null there, reducing the need for PEQ at that frequency. Whether you high-pass the mid/highs or not, there is still going to be a defined acoustic crossover point. Not time aligning at that point plus overlap is creating at least some of your room problems.

Dennis

Gadget

The minimum delay, filter or no, is .02ms...As Dennis pointed out WITH the mid/highs on the 260 you have so much more control... I use one on my Infinity RS1's and Sunfire subs... and its very transparent..You should hear the difference with the 4800 though.. if you want transparency.. THAT is THE unit... :shock: but spendy...but MAN do you have control with that unit...but I digress...

Hey it doesn't cost anything at all to try it both ways...Have you experimented with speaker location? I took DAYS finding the best location for my AC5's in the sudio...

Remember that the LF is subject to boundary cancellations as well...Have you run a room mode calculation? What are the principle modes?

I guess what Dennis and I are saying is ...try the system with the 260 in total control, and with it just processing the low's...Use the speaker placement and Room mode calculations to make some processing decisions...

I have also used delays to compensate for room modes, or for steering them to area's less critical to listening...

Gadget

Dotsystem

Guys

I have the BARE software and XTZ software/hardware. Both of these sweep and compute the PEQ settings. My main problem is around 40 hertz.

With another speaker I tried running my system through a Rives PARC, a very expensive PEQ marketed to home audio types. This was claimed to be totally transparent. Problem is, it is not and it could be heard having a slight masking effect at frequencies way above it PEQ settings.

With regard to running the whole system through the 260, I will rely on my past experience and the reported experiences of other home audio types using various brands of pro-audio gear. As far as relative sound quality, the 260 is the weak link in my system but the ear is less critical with the bass frequencies. I can hear some loss in signal quality without any filters, gain change, etc. engaged but this is still better than the room boom that occurs without it.

From where I sit, no good result can come in the midrange or treble from feeding a high resolution analog signal already converted with a 32 bit DAC from a 96/24 or 192/24 source though an additional ADC/DAC chain and outputting a 48 derived analog signal in its place. I expect I would like that less than the Rives PARC result. I can also understand how using the 260 to control the full bandwith is a totally acceptable approach to many. Home audio can have some diffferent considerations and using the 260 for full bandwith control would be OK up to a certain level of system quailty. Based on my Rives PARC experience, I think it would be a step sideways for me at best and a step backwards at worst.

Some enthusiasts have used fancy and expensive DSP room correction systems and eventually evolved back to simpler scenarios with minimal signal processing/more room treatments. In my case, massive amounts of treatment would be required since 40 hz is pretty low to absorb.

Bob

Dotsystem

Gadget wrote:

Have you experimented with speaker location? I took DAYS finding the best location for my AC5's in the sudio...

Remember that the LF is subject to boundary cancellations as well...Have you run a room mode calculation? What are the principle modes?

I guess what Dennis and I are saying is ...try the system with the 260 in total control, and with it just processing the low's...Use the speaker placement and Room mode calculations to make some processing decisions...

I have also used delays to compensate for room modes, or for steering them to area's less critical to listening...

Gadget

I have experimented with speaker placement quite a bit, use laser levels to set the tilt and convergence, distance meter to equalize the distance. There is a trade off beween frequency response and perceived image quality with each pair of speakers I have tried.

I have noticed that if I add a delay to the bass (therotically making things worse), this does have a pleasant effect on some recordings. One thing I need to explore is extending the time window when I make my sweeps to see if I can improve my PEQ settings.

Perhaps I will try it full range with low expectations. The high cost of a better Driverack would inspire me to research all the viable alternatives.