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.

Thanks

Barry
 

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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.

Do not:
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.
 
BesPav,

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.

Barry
 

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Hi,
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?
 
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Hi krivium,
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
 

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Hi,
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.
 
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All working

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
Barry
 

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Yes i used ADAT and still use it ( a lot! ..as one of my digital mixing desk run 24/7 at a friend's webradio ( Yamaha 02r to Rme Adat card )). In general i like the optical interface ( spdif too).
It is reliable and sound as good as aes/ebu ( self clocked). It solve many issues regarding groundloops ( if you have a piece of gear with analog involved). They are visually inobstrusive which is nice too.
I don't use it with my Lake as it is aes/ebu oriented but if availlable i would happily do it.
Be careful with cable though, they must be good quality. I'm not saying fancy gold plated thingy i talk about professional gear used in studio.
Check with studio dedicated shop rather than hifi or audiophile.
 
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I'm using two different filters, one for listening to music & one for watching blurays.
Music one is FIR, with LR4 crossovers & phase for each driver flattened which if I try to watch blurays has ~ 0.5 seconds delay.
Bluray one is just a basic crossover LR4 slopes, but no time delay.
Hence I was thinking of using optical in from bluray player and then make the filter more sophisticated, need to buy an optical cable
 
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^ ok thank you for clarification.
About Adat and blu ray it didn't occured at first to me but are you sure the bluray have ADAT out?
It could be the case as it could output 8 channels on one optical fibre those being interesting for multichanel setup ( 5/1 or 7/1, whatever...) but not so much for music or LR.
It will work with 2 chanel no issues but could be a waste and spdif optical ( toslink) could make more sense from a brand point of view ( economy).

Anyway if you plan using optical interface to compensate for delay you will be disapointed as most delay with FIR comes from the treatment itself. In other words you can't go back in time.... but you could do the other way around: hold for a fixed amount of time ( equal to your overall audio treatment latency) your 'image' video message.

This is availlable task in something like Jriver. I don't know what is your plan longterm for pc/filters ( stay with reaper and it's deconvolution engine?), but there may be a dedicated video plugin or function included in reaper to do same thing.

If available you could have something like blu ray hdmi to pc hdmi input card to enter everything into reaper or Jriver.
If your tv have hdmi out you could do the same.

In the end you could use FIR preset for everything the only drawbacks being you'll need some other gear to acquire ( but hdmi input card should not be this costly i presume), you'll have an overall delay to your system: it will respond slower with a delay equal to the latency needed for the FIR.

The other answer is to forget bluray, invest in some ( external or internal computer) digital media readers ( bluray, dvd rom, cd rom) and connect them to the pc running reaper, use Jriver as global player ( audio and video in place of bluray) and use an audio loopback driver ( vincent burel 'audio cable' comes to mind -freeware) to input audio in reaper.

This is one scenario, others can be thoughts of...( check vb audio soft offer for some inspiration).
In the end it'll depend of your needs and preference. This one of the strength of computer based system. But a weakness too as being completly open it is easy to get lost or sidetracked by options and possibility.

About your filters you could use steeper crossover slope with FIR there is no drawbacks from the filter algorythm themself ( except latency). My experience with direct radiators/ tweeter is that it help to limit drivers acoustic overlap and then limit lobing issues.
It can help. Or not. As other thing can come into play... :D

I've never played with an horn loaded system like yours and FIR but i bet it can be interesting: make a nice step response by time adjust the drivers ( and eq them flat individualy) and optimize the xover freq for good directivity match and lower lobing behavior i would expect to be very rewarding.

But i drift to subject which should be in the multiway forum.
 
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Thanks for your helpful comments. I continue to learn. Here is me thinking - bluray has toslink connector, so does Apogee (but it's called ADAT), connect them together and expect them to work:confused:. I understand now that this is not possible.
I can still use the analogue out from BD player & analogue into Apogee.
There are several plug-ins in Reaper which I can explore / play with - a declick for LP listening.
I'll continue listening and playing with this arrangement.
JRiver was a consideration before this PC & Apogee came along and could be used in the future.

Again, thanks for taking the time to explain options.:)
 
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Well no this is not possible to connect optical connectors and expect this to work without checking the kind of transfer protocol: multiple transfer protocol use the same kind of conductors ( optical fiber) but are not all cross compatible (eg: adat and spdif/ toslink are not , but aes/ebu and Spdif ( wire) can be ( with some additional gear).

The Adat on Rosetta is here for multichannel purpose mainly ( these units often replace analog taperecorder in studios) as Adat is a widely used interface in pro world.

I think it is wise for the moment to use the bluray as an analog source ( the time you get used to the whole installation): the Rosetta AD sounds good at 96khz and you won't loose too much in quality if Bluray converters are ok.

It is possible to transfer bluray audio in realtime digital to your pc for sure but it'll probably make the system use a bit tiring (at least at first):
If you have a mixed use of bluray and cd you'll need to make 2 more sets of filters at 44100hz and 48000hz and load the one needed depending on source as it could introduce conflicts with the setup you have dialed now in reaper ( 96khz).
So you'll have to load differents projects in reaper (eg: cd and video) and change the clock source ( the Lynx and Rosetta slave to bluray digital stream).

Theorically the clock signal will be of lesser quality than Lynx's and/or Apogee's clock but the Apogee is supposed to re condition the incoming wordclock for the better ( it has a dedicated reclocking circuit iirc) and there shouldn't be this much difference in the end.

An other solution could be to use a realtime resampler but it will induce other tradeoff: hardware is going to be costly (Mytek had a 'good' sounding one for something like 2000euros...), software could need a bit of audio routing and will induce another source of latency.

Another solution could be to use another DAW than Reaper: Sequoia was able to load and play different kind of frequency at the same time...

There is this is sure other options possible... only your way of using your system can tell which one is for you.

About vinyl i don't use realtime plug ins for treatment. I first digitalize through Lake converters then i treat them offline using Ozone RX ( pro cleaning dedicated sofware). I then store the files and play them directly from SSD HD.
Saves on my precious mc needle life and allow High Definition recording of vinyl ( 96khz/24bit as you :) ).

I found that keeping physically rotating device at minimum in my pc ( only cooling fans and psu / semi passive) made difference to quality ( so SSD for storage).
 
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