Apogee Rosetta 800 +
I've built my 4 way horn loudspeakers and been playing with REW & a Behringer DCX2496 to establish crossover points, slopes, time delays, etc.
The opportunity to buy an Apogee Rosetta 800 192kHz has come up and I'd be grateful for comments on my proposed system configuration.
Majority of music listening is from NAS drives, but I'd also like to be able to listen to vinyl, blurays & TV.
I think miniDSP is the weak link, but all suggestions welcome.
Fabfilter Pro-Q ?
Thanks will have a look.
No adverse comments from anyone to proposal, will proceed once we can arrange a visit post Covid-19
You're now in a very strange situation.
First of - you need a good filtering properties. Better if capability to use FIR compensation for each band. Even better if capability to use calibrated microphone based optimisation.
Next - you need to use masterclock-syncronous 8-channel DAC.
Next - you need to use 8-channel volume regulator, because your system pretend to higher end and dropping of some bits must be inacceptible.
So there are some different ways to achieve those goals.
First, agree, Rosetta. But you'll need to find comprehensive (and either anyway overpriced software or dive into manual of brutefir) filtering software and build 8-channel analog volreg.
Second - use something like Symetrix Symnet 8x8 or DBX Driverack 480/4800 units. Then you can implement FIR eq in a group stereo signal from PC and next divide it to bands via such a units.
Even try nanodigi, because it provides four stereo SPDIF signals which you need to DAC from one masterclock.
Even try a pair of DBX 260 or Behringer DCX2496 because they will use different masterclocks.
Even try to volreg signal before last DAC chain link, you 'll greatly sacrifice dynamic range.
Many thanks for your comments, they are very helpful, especially the do nots.
I found a review of minidsp U-DIO8 on audiosciencereview.com which made me look for other alternatives. The seller of Apogee has a PC with Lynx aes16 card which he going to sell me as a bundle (see revised layout).
He uses Reaper as a VST host to do all of the crossovers.
Initially I'll do volume control in software and then in the future look at building a 4 channel stereo vol pot.
I can't follow Bespav comments about a masterclock ( which isn't needed as long as you use your last scenario and without other digital gear within your signal path) neither the 'difficulty' involved with software for filtering: any Vst host can run a multichanel convolver reverb plugin ( voxengo 'pristine space' is an example) in which you can run a set of filters created using Rephase ( IIR or FIR).
In fact i run more or less the same 'scenario' as your last one in my main system ( Rme Aes Pci to Dolby Lake /FIR filtering) and the one i described using 'Pristine Space', an older 'pro' soundcard (Aardvark Aark24) on an old pc ( running under XP) in a secondary system.
Both works correctly ( if you can withstand the latency induced in both case).
The Rosetta will let you calibrate internally ( using a screwdriver) the analog output level ( +4dbu or -10dbv).
Bespav comment about the need for an analog attenuator between your DA and amp is true. An ideal solution would be a 'R2R' stepped attenuator ( eg: Amb lab delta 1) but it may be complicated to find a 4 way balanced capable unit.
I run an hybrid approach in my main system: i've got a three preset analog attenuator between the DAs and amps ( with high spl, mid and late night position) and i use the digital input attenuator in the Lake to fine tune level). It is a bit more complicated than a multichanel steped attenuator but satisfying in daily use to me ( and as it is intimidating to my family it keep them away from this gear... they prefer my analog chain which is simpler to use).
For the secondary system i use the Vst host internal ( digital) volume only.
Your system will sound good with this Rosetta, i always liked it in studio. Not the most transparent DA on earth but very nice sounding ( in fact to me it was the better in Apogee range at that time). What do you use as DA at the moment?
thanks for your comments.
Is your comment on latency related to watching video whilst playing music or to do with music?
Your hybrid approach using a 3 position attenuator sounds an easier proposition than infinite adjustment.
I'm encouraged by your comments on the Rosetta.
DA at the moment is IQAudio DAC on top of the rPi. This goes through a PureSound L300 preamp -> Behringer -> amps. I had a passive 1st order crossover to JBL 2405. see attached.
You can see why I want to change.
Before I built these horns, I was using either Heco Direkt or Living Voice Avatars with Puresound M845 amps
About latency: either case ( music or image) it will happen.
There is multiple source for latency: your AD and DA introduce some but it is usually very low and negligible ( 1/FS each time there is a converter so if AD and DA used in the chain the total latency induced is 2/FS).
Then you may have some computer induced latency ( depend of the driver, the hardware and software used). This is less easier to define as it will depend of your configuration but as you'll use pro gear ( Lynx and Rosetta) it should be low. Maybe 3ms in worst case but i'm pretty sure you can halve that maybe even lower.
Then you'll have treatment latency ( for your xover, eq,...). Here the problem will be dependent of the kind of filter used ( IIR or FIR) and the optimisation of the soft used to perform them. FIR will be the worst regarding latency and here again it'll be dependent of a number of factor ( steepness of filter and frequency, the lower the freq the higher the latency the steeper the more latency).
With music except if you do real time monitoring ( iow if you record a musician playing while listening with your system) it should not be a problem but in case of image it can quickly be an issue as we human can be disturbed with latency as low as 3/5ms. So you may have issue when watching tv or playing a bluray.
That said nothing stop you to have different settings for different purpose: an IIR set up for tv/bluray (trading 'accuracy' for latency) and a FIR for music or critical listening.
To be honest this is what i've done as i used my system for tracking with musicians.
I don't know if i was clear enough in my previous answer but the point was that a software solution for xover treatment is perfectly doable even with freeware and older computer. You don't need a dedicated loudspeaker management system ( even if it has some advantage to it).
About your config of the moment you have multiple stage of conversion and the probably 'worst' DA in your chain are from the dcx. Expect a real gain in quality with the Rosetta. It may not be obvious at first but you'll soon find the dcx poor once used to the Apogee.
Lynx is very nice card, rock solid stable and with good clocking. There is low chance you'll be disapointed with this gear.
The 'hybrid' attenuator is an handy compromise: you don't sacrifice too much regarding number of bits wasted in digital ( in my case i rarely use more than12db digital attenuation and to be honest this is not something that bother me to much and an 'academic' concern more than real life) and can taylor easily the attenuators ( Lpads) to your needs. And it is simple in practice: a bunch of resistors and relay, some diode to protect your psu from relay kickback, a Lorlin switch and small smps is all what is needed.
That said you'll have to define clearly your requirements about levels but you shouldn't have issue doing it as you seems very well organised and meticulous in your approach.
Many thanks for the explanation and suggestions. With luck in early Aug, I'll be able to arrange for the kit to be brought here and set up.:)
Apogee + PC installed, all working. :D
Initially with just a local hard drive, but more recently with NAS, then Bluray player.
Edd (who I bought the stuff from) very kindly individually measured each driver in REW, tweaked them in Rephase before loading the parameters into Reaper (FIR filters). We're using the clock in the Apogee as the master & running everything at 96/24.
Main listening is using Orange Squeeze on a tablet to control LMS with music from the NAS.
As Krivium suggested, I've had to compromise, due to latency, to watch Blurays, but they still sound good.
Volume control is software at the moment.
Thanks for the help
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