TOPIC:

domin wrote:

You make things too complicated!
It's correct but doesn't correspond at a regular system...

A Nanoavr is hdmi.
The only way to process bassmanagement before the Nanoavr is too have a PCHC as source (and the soft to do it) or use a Nanoavr HD...
Not a standard configuration , but more a complex one!

The Nanoavr must be before the AVR (or preamp) because after that, there is no hdmi signal...
So the configuration is thus after the Nanoavr.

You can use a nanoAVR HD ahead of the nanoAVR DL (what I do), or you can get a DDRC-88 with DL and the BM plugin. But, with the DDRC you're either stuck with an additional A/D-D/A step or some way to get AES3 digital audio from your HDMI source. A lot if people use DDRC-88A after a preamp. To get AES3 audio from HDMI you can convert to SDI and use a deembedder.

Using just a nanoAVR DL involves a compromise which might be reasonable for the price for many people.

I was looking at a DDRC-88D but scored a nanoAVR DL from eBay so am going a different route: OPPO 103D to nanoAVR HD to nanoAVR DL to HDMI to SDI converter to SDI digital audio deembedder to dual AJA ADA4 quad DACs to smps and speakers. It's still cheaper than a DL pre/pro. A DDRC-88A would be cheaper still but I wanted to avoid the extra A/D D/A step and do as much processing digitally.

domin wrote:

You speak about the software for PC.
It's not the same placement than a Nanoavr !
So configuration it's not the same process

Sigh. Yes, you can implement all of Dirac measurement, computation, and filtering in a PC. Here, the PC measures, computer, and then transfers FIR and IIR filter parameters to the nanoAVR DL.

Same caveats apply.

tecnogadget wrote:

This is exactly where I wanted to get.

If i have to put a nanoAVR HD before an nanoAVR DL this means is a terrible product. Like there is no point at all buying a nanoAVR DL if its not going to do the job since they “could not fit the needed processing power in the DSP to handle BM”

Following that reasoning one may conclude the Hdmi AVR DL is useless, not a complete solution and the real product is the DDRC-88 (twice the price $)...im not making an statement, just saying what people may think.

The hole thing is so confusing that at the end of the day you turn out spending on a AVR that has Dirac build in just because you dont want to mess up with BM.

Don't misunderstand me, i want to believe in the nanoAVR DL, its supposed to make pEQ automatically for surround sound at a kinda reasonable price...but the do after/before BM think is quite misleading.

I think this topic should be clarified better in the official webpage/brochure/manual/Faq to stop the endless mistery for the users.

The nanoAVR is not useless. It is limited. FIR filter procesding is computationally expensive compared to IIR. A nanoAVR DL and a pre/pro that does no correction is better than, well, no processing.

You correctly note that it is half the price of the DDRC-88 series, and, not unreasonably, not as powerful. For many of us Dirac in a small box on the digital domain is very appealing.

The areas where we debate BM before and after DL really are perfectionist nits, at least as they apply to theatre. I think BM after DL in a properly level and distance calibrated pre/pro is fine at the sweet spot. Yes, you are mixing channels downstream of DL but with a proper movie mix there should be little low frequency signal in the non-sub channels. Stereo music may be another issue but again I think the benefits may outweigh the disadvantages for most theatres: You're already crossing to a sub - how much worse can it get? Level and disyance adjust in the pre/pro and you shouldbe fine.

The nanoAVR alone will improve a non-corrected system, even if not optimally: better does not beat perfect, but it beats nothing.

It is optimal - or at least, exactly the same result measurably and audibly as a well-configured system with the same BM before the DL. That is assuming a straightforward 5.1 or 7.1 system (or stereo), and that the BM in the AVR is good. Where it gets unstuck is weird things like Prologic and not having a center channel.

john.reekie wrote:

It is optimal - or at least, exactly the same result measurably and audibly as a well-configured system with the same BM before the DL. That is assuming a straightforward 5.1 or 7.1 system (or stereo), and that the BM in the AVR is good. Where it gets unstuck is weird things like Prologic and not having a center channel.

There you go. Take John Reekie's opinion over mine any day.

rhollan wrote:

john.reekie wrote:

It is optimal - or at least, exactly the same result measurably and audibly as a well-configured system with the same BM before the DL. That is assuming a straightforward 5.1 or 7.1 system (or stereo), and that the BM in the AVR is good. Where it gets unstuck is weird things like Prologic and not having a center channel.

There you go. Take John Reekie's opinion over mine any day.

I wonder of Mr. Reekie can elaborate on that. I'm guessing that mixing bass signals after Dirac live isn't all that terrible a thing to do if the levels and delays of the individual speakers have been matched in the AVR or pre/pro. Even if not, I'm guessing that Dirac Live can try to level match by way of response adjustment and delay match by way of impulse correction but this is not what it is designed for. In particular, trying to delay match by playing games with impulse response may result in FIR filters too long to accommodate the delay differences between drivers if they are too far apart already.

I ran my first set of measurements with a nanoAVR and Dirac Live yesterday! This was not the final configuration, less than ideal, only roughly level and delay matched, but I was impressed at the corrected response curves it came up with. Though this probably deserves a topic of it's own, I'll describe the setup here, for starters.

Right now I have an Oppo BDP103D BlueRay player doing level and bass management out of it's analog outputs, going into an Outlaw 990 that does nothing other than master volume control and convert unbalanced to balanced signals. From there, two front BG Radia 520 speakers are driven by a Crown XLS 1502 amp, one BG Radia CC-220dx center speaker driven by an Outlaw Audio M-2200 monoblock, and two non-descript small rear speakers driven by a Crown XLS 1002 amp. Two dual mono Klipsch 12" subs complete the mix, which runs 7.1 mixdown to 5.1. I am waiting for four BG Radia PD-6 LCRi speakers to arrive to serve as the rears, and they will be driven by Crown XLS 1002 amps, in pairs.

The room is small, 13 feet wide by 14 feet long, with an 8 foot ceiling. Front right has the entrance door, and center left has a walk-out patio door, covered by a blackout curtain, The room is carpeted, but rather acoustically lively. Subs are in the front, about 18 inches from the wall, at the 1/3 and 2/3 positions across the wall (4-1/3 feet and 8-2/3 feet). Front speakers are just to the outside of the subs, and the faces are aligned. Center speaker is, well, in the center, just below a 52" 1080p TV. Rears are directly opposite fronts, and sides will be on the sides, about three feet from the back of the room. Seating is a single three seat sofa four feet out from the back of the room. I do have a set of four rear/surround speakers I could use for 7.1 instead of 5.1 but currently am not.

My testing involved just running the HDMI signal from the laptop through the nanoAVR DL, thence the Oppo 103D (It has two HDMI inputs that it can process as it does the content from a disk), via the bass managed (and poorly) time and delay aligned analog outputs to the preamp acting as a master volume control, and thence the amps and speakers. The same laptop was connected to the UMIK-1 measurement mic and the nanoAVR DL via USB. So, this is a BM after DL configuration. After some level tweaking (the Crowns all have attenuation controls and the Oppo BDB-103D offers level trim as well), I managed to get some measurements, and could compute Dirac Live IIR and FIR filters.

Wow! I expected the close proximity of the subs to the front speakers would make for easy impulse response correction, and it did. Impulse correction for the rears was less stellar. But, this is understandable: the subs are some ten feet away from them and bass was being sent to them. What was surprising was the corrected frequency response of the rears (remember, these are channels, not speakers, what with bass management downstream of Dirac Live): a beautiful corrected flat line down to about 20 Hz. I expect the rears alone are 6 dB down at 120 Hz, given their size). The front channels, by comparison, had great corrected impulse response, but the frequency response below 60 HZ was a bit wonky (some phase correction between the subs could probably help here): a bit of a 5 dB hole from 40 to 60 Hz. But, overall, much better than uncorrected. Do realize, this is about the worst setup downstream of Dirac Live one might have: rough time, delay, and level alignment.

Next steps are to integrate the new rear speakers (They are in-walls so there will be much cutting of drywall and consternation involved), and replacement of the Outlaw 990 via analog connections with an HDMI to SDI and SDI audio de-embedder setup, once a nanoAVR HD arrives to do bass management (and level/delay adjustment, if I put it downstream of the nanoAVR DL). At least I can experiment with BM before AND after DL.

I have to correct myself here: non-sub channels are full range and it is up to the reproduction end to direct bass to the most appropriate driver, generally a single or dual mono subwoofer setup. The important thing is that the drivers for a given channel are sufficiently time and level aligned downstream of DL processing so as to not overwhelm it. IIR filters can compensate for mismatched levels between drivers, to a point. FIR filters can compensate for delay mismatch, again, to a (much smaller) point. But that is not what Dirac is designed to do.

I'm not sure why mixing Dirac Live corrected channels downstream is such a bad thing. After all, the reproduced sound mixes between the speakers and our ears. I can only presume this is the case if the intent is do clever tricks with phasing between channels, which Dirac Live will almost certainly alter: phantom centers and Prologic decoding. (John Reekie alludes to this, and I would appreciate clarification if my understanding is correct).

The advantage of doing bass management downstream of Dirac Live processing, of course, is that Dirac can make corrections in the crossover region, though this should be unnecessary for well-designed crossovers (like 4th order Linkwitz-Reily) if they appear before Dirac Live processing. The only caveat I can see is that it is best to delay and level align drivers downstream of Dirac Live if doing bass management after it.

I suppose it all comes down to what you want to correct: individual speakers, or speakers in multiple enclosures combined together (and therefore their combination -- which happens within two and three way speakers anyway).

I've found this data searching for a diferent topic about the interaction between Dirac and upmixers like Dolby Pro Logic IIx and the updated version Dolby Surround (DSU)...

This is a repost from Devteam , original posted by Tony at AVS forums

Bass management + Dirac: A very interesting and lively topic! Glad to see all this brainstorming going on
OPTION1: Source - > Dirac Live -> Bass management (BM) - > AVR (nanoAVR DL structure)
OPTION2: Source - > Bass management(BM) -> Dirac Live - > AVR (proposed as the ultimate solution by some of you)

Knowing Bass Management is just a glorified Matrix + Crossover network, OPTION 1 or OPTION2 is similar to EQ in a multiway active system . Say a 3way stereo system where you have EQ at the input side(for L&R) but also have EQ at the output side (for each specific driver: Low/Mid/High). Would you agree? My point being, this EQ argument isn't new! it's been there for many years. There will be 2 sides of the argument, both with very valid arguments, both with Pro's & Con's… The perfect system (to my mind) is actually a combination of both worlds, this way everybody is happy!

OPTION1: Source - > Dirac Live -> Bass management - > AVR (nanoAVR DL as of now)
This is similar to doing EQ at the input section of an active system (i.e. L&R overall EQ). When we run 8 measurements at 9 positions = 72 measurements, we're building a "model" of the system. In other words, we play a sweep on Front Left (as your bluray player will when content is played), it will indeed start sweeps @ Low Freq inside the SUB (routed thanks to BM) and then move to the Front left pass cut off frequency.
PRO's: That's the true response of the system (i.e. extended bass response inc potential crossover effect). Dirac will not only fix the speaker but also the transition to the SUB. It's easiest way to fix your content (i.e. 7.1 or Stereo) no matter the configuration you have.
CON's: The impulse responses may be hard to work with however as the sound is arriving at different time for different frequencies in different positions (the sound is not coming from roughly one point in the room) and if the delay and level differences between a sub and a "main" speaker is very large, it may be difficult to handle within a "combined" Dirac Live filter. Now, if you run Dirac Live trial on your PC/HTPC, you're most likely running under that configuration (AVR doing the BM). So have a trial at it. It's free to test and you'll see the results..

OPTION2: Source -> Bass management -> Dirac Live @ each output- > AVR
This is similar to doing EQ at the output section of an active system (i.e. each speaker/ each driver / so speaker EQ). You'll build the best response for each speaker. There are 2 possible locations where we could run the sweep signal for Dirac measurements in this configuration:
a) Sweep before the BM module to give the true color of where the signal is routed to = > i.e. sweep will go through Sub + Left speaker (thanks to BM) PRO's you do see the true colors of the entire system. CON's: you're now planning to apply correction to only one output since Dirac is "post BM" (i.e. DSP block at the end of the chain). It will not solve your problem in your room. (measurement and filters are not in sync of what's to fix). This solution isn't right
b) Sweep after the BM module to give the true color of how each speaker sound =>i.e. Sweep will only go through the speaker.
PRO you will do GREAT at solving your speaker. Produces the best possible speakers for the bass management to work with.
CON's. what about all the filters you may have in your BM section? They are not taken into account. i.e. once you do have BM enabled, the sound will change so your measurement isn't a representation of the system. Possible your bass management crossovers will create a dip/bump at the LFE transition. Dirac will unfortunately not be able to fix that. (post BM, can't see it).
The way we see it, Option2 will work very well if you know really well what you're doing upstream (i.e. at the BM level). Call it an "advanced mode". The flexibility of our BM module as shown by Kal's extensive review on Stereophile is a testimony of what you can do once in the right hand. With this said, we've got to be realistic that it's not as plug&play. REW, filter and acoustic interpretation of results will take time. That was the bottleneck which we evaluated at length, and with our objective of putting Dirac Live in the hands of the largest crowd (Affordable/Simple/easy to use), we tried to make it an accessory to an existing system that is most important plug&play. Don't get me wrong, Option2 can be a great fit too for complex setup and I'm sure that a lot of tinkerers with complex system would love that. I just hope that it makes sense to you that it's maybe for a different crowd. Maybe a different product. Different philosophy.

That's all for my "little" addition to this topic and sorry for the wordy reply... Tried to include all my thought in one shot, most likely missing some but hopefully showing that each argument has merits.

Tony

I've read that, technogadget, but thanks.

My concern is why does Dirac suggest mixing before and crossing over after Dirac Live correction. Two reasons that I can see are

1) DL correction will mess up phasing between channels so decoding things like Dolby Pro Logic or using phantom center speakers won't work.

2) To work properly, DL will require the drivers making up one channel to be time and level aligned. This is one more thing to "get right". (And, indeed minidsp's own nanoAVR DL manual mentions this, presuming the pre/pro or AVR will be downstream of DL.).

So, if you get (2) right, and (1) is not an issue (you are mixing unrelated signals: bass in LFE and bass in a non-LFE channel), then BM after DL is "fine", yes? We all agree? Experts chime in please.

Now, the purported advantage of BM after DL, is that DL can "correct" for crossover beween main speakers and subwoofers, where if BM occurs before DL, it can't. O.K. I get that, but what is there to correct?

Especially if you know the characteristics of your BM crossover, as you would in, say, a nanoAVR HD, why does this matter? The crossover should be well-behaved, and understood, and if you insist you can certainly (and probably should) use something like REW to measure the combined system. 4th order LR filters, for example, are constant amplitude (though REW generally recommends 24 db/octave drop on the high side of the low pass filter, and 12 db/octave on the low side of the high pass filter to match the existing 12 db/octave likely rolloff of the high frequency driver at the low end for a combined equal amplitude response). Why measure the BM crossover at all as part of DL measurement?

The only thing I can think of is phase response through the upstream filter affecting measured overall impulse response, and DL measurement won't "see" it. But for 4th order LR filters the phase difference between high and low side is zero. However, the phase response is not linear, and thus will affect the impulse response of the system as a whole, no? That would suggest either using a linear phase bass management crossover before DL, or a more traditional Linkwitz-Reily crossover after DL that "sees" time and level aligned speakers.

Comments from the experts?

My gut tells me it's easier to let DL measure distances and levels to individual speakers and then do BM mixing and crossovers before it, because those are rather set and forget (80 Hz here, 100 Hz there, etc.) But if you can't do BM before DL, make sure your pre/pro or AVR blends the disparate speakers together (sub and sats) as well as possible.

However, I think that DL can also correct for phase non-linearities in crossovers (such as within individual speaker cabinets, as well as between speaker cabinets as part of BM) as part of impulse response correction, and so would benefit from "seeing" such crossovers provided that separate speakers being blended together are level and delay matched.

Am I getting the jist of it, or am I way off?