[audiogurujax]

OK I am building my dream speaker an array with 8 Neo10 and 24 Neo3PDR per channel, the system will be active I currently have 2 PrimaLuna Dialogue Premium HP amps and 2 Dsonic M3-600, I have 3 options use the PL's in stereo and have one amp for the Neo10 and the other for the Neo3 or run the PL's in Mono and use them for the Neo3 and have the Dsonic run the Neo10 or invert it and Have the PL's drive the Neo10 and the Dsonic the Neo3 I will be crossing around 3-4 Khz so everything between 150hz and 3-4Khz will be on the Neo10 thats why I am leaning to use the Tubes amps there and let the Highly detail Dsonic run the High frequencies.

What you guys think?

[mlundy57]

Any reason why you can't try them each way and see which you like better?

[Danny Richie]

I think that is WAY too high of a crossover point for those drivers.

Also, consider the impedance loads that these drivers will be in series paralleled groups. The tube amps my or may not like the impedance loads depending on how you wire them.

[audiogurujax]

My experience with the PL is that is sound the best using the 16ohm tab followed be the 8 then the 4 which I always feel is bright. So I will experiment for example 8 Neo 10 can be wired in  2 parallel sets of 4 in series (16 ohms apx little less resistive) or 4 of 2 (4 ohms) with the neo3 if I use 21 of them I can wire them in 3 sets of 7 in series I think that yield close to 10 ohms. I think for the PL amps it is important to keep the impedance higher than 6 ohms. Not so much for the Dsonic.

Danny in terms of crossing frequency what do you recommend 2-2.5khz?

Thanks a lot,

Eric

[audiogurujax]

The reason I want to use the PL in mono is not because I need the power, is because, and I do not know why, these amps sound great in stereo but when you run them in mono they became one of the best amps I have heard in my life, they sound very different, a huge improvement in every aspect. I asked Kevin from Upscale and he said it could be due to having dual output transformers and something regarding the input impedance, not sure LOL

[Danny Richie]

Danny in terms of crossing frequency what do you recommend 2-2.5khz?

I never decide on the crossover point prior to taking measurements to see where the natural roll off of the drivers are. But having worked with those drivers in the past, and running 21 Neo 3's in a line, I can tell you that depending on how they are mounted and how they are rear loaded you should be about to cross them in the 1,100Hz to 1,300Hz range pretty easily.

[audiogurujax]

Interesting isn't better to have the same driver reproduce the whole critical audio range (300-3000 hz) because the neo10 can reach 3000hz easy what is the advantage in crossing lower? Sorry you are the expert and I am the annoying student asking a million questions lol!

[Danny Richie]

Interesting isn't better to have the same driver reproduce the whole critical audio range (300-3000 hz) because the neo10 can reach 3000hz easy what is the advantage in crossing lower? Sorry you are the expert and I am the annoying student asking a million questions lol!

That critical mid-range is 300Hz to 500Hz with another octave above that for mid-range overtones and higher frequency range instruments. So really 300Hz to 1kHz is your heart.

Also, if you cross at 3kHz then the two sets of drivers will cancel each other out in the horizontal off axis response. So you'll have holes in the off axis.

And, as good as the Neo 10 driver is it is still no match for the Neo 3's in ranges above 1.5kHz.

[Tyson]

Danny is right on.  Since you are active, you can try it out yourself.  Set the midpanels to go from 200hz (or 300hz) to 1khz and have the tweeters take over at 1khz.  Now, mute your tweeters and listen to just the midrange doing 300-1khz.  Sounds pretty good!  Now unmute the tweeter and mute the midrange, so now you are listening to the tweeters only from 1khz-20khz.  Sounds like very little.  Mostly just overtones and ambient information. 

Also, since you are active, you can move the crossover point around (higher or lower) and see what the effects are.  But IMO between 1khz and 1.5khz is about right especially if you have that many tweeters. 

[Captainhemo]

It'll be interesting to   get some measurements and  see just how low that row of tweeters is playing... go from there.  You've got a nice guideline being Danny has   used rows of Neo3's in the past but  a measurement   still goes a long way

jay

[audiogurujax]

Question, I have 22 Neo8 PDR drivers at home will it be comparable the sound if I use 10 of these per channel in place of the Neo3 PDR (8 Neo10 and 10 Neo8pdr per channel crossed at 1khz) because if the performence wan be similar why spend over $3,000 on tweeters, I will do that only if the sound and performance from the Neo3 is very substantial.

[Danny Richie]

Question, I have 22 Neo8 PDR drivers at home will it be comparable the sound if I use 10 of these per channel in place of the Neo3 PDR (8 Neo10 and 10 Neo8pdr per channel crossed at 1khz) because if the performence wan be similar why spend over $3,000 on tweeters, I will do that only if the sound and performance from the Neo3 is very substantial.

Actually the sound is pretty close. The Neo 3's are flatter in their response and more extended above 15kHz. But that is about it.

[audiogurujax]

So since I am in my 40s and I can't hear nothing above 17khz I should stay with the neo8 lol and the flatness I can correct it with the DSP! Still 21 neo3pdr look really badass.

[Tyson]

I ran dsp active for a long time and after all of it, my experience is that you should only use DSP correction as a last resort.  The more you apply, the more it degrades the sound.  Not sure why it's true, but it is.  I wish it weren't, because then I'd still be running active.  But in the end, get good drivers with a flat response naturally, use a simple and low order crossover and keep your impedance as flat as possible.  That's the key to good sound, IME.

[Danny Richie]

I ran dsp active for a long time and after all of it, my experience is that you should only use DSP correction as a last resort.  The more you apply, the more it degrades the sound.  Not sure why it's true, but it is.  I wish it weren't, because then I'd still be running active.  But in the end, get good drivers with a flat response naturally, use a simple and low order crossover and keep your impedance as flat as possible.  That's the key to good sound, IME.

I wish I could get people to understand that. Everyone thinks it must be the newest, latest, and greatest way to go and it gets all of those passive parts off the drivers.....   But in reality it just doesn't sound as good.

[Folsom]

I ran dsp active for a long time and after all of it, my experience is that you should only use DSP correction as a last resort.  The more you apply, the more it degrades the sound.  Not sure why it's true, but it is.  I wish it weren't, because then I'd still be running active.  But in the end, get good drivers with a flat response naturally, use a simple and low order crossover and keep your impedance as flat as possible.  That's the key to good sound, IME.

I have yet for DSP to blow me away. I remain open minded about it, but it has to prove it's worth to me. Any setup I've heard with it wasn't really notable for the DSP aspect.

To me people look in the funniest of places for everything they expect to improve a stereo.

Jax, I hope pictures are coming when you're done.

[audiogurujax]

You bet I will post pictures, I am friends with 2 recording engineers and they always making fun to me, because they say you audiophiles don't want to equalize or DSP but you don't realize that we equalize and DSP the hell out of every recording we work on and most of the best studio monitors are Digital and use DSP, lol I think he have a point. My personal opinion is that yes there is something special about analog! But I got tired of spending thousands equalizing my audio system by changing cables, replacing components or by buying the latest tweaks, is a very expensive and unpredictable way to fine tune your system, you want to fine tune or equalize your system buy an equalizer not 10 different pairs of interconnects lol.

[Folsom]

Ya but most new music doesn't have many of the qualities found on analog recordings. And they forget that their DSP/eq'ing isn't done in real time, it's done on really high resolution files and takes awhile to apply to the tracks.

Besides despite all their meddling, the overall product may come out fine, and sound a lot better than on most engineerings speaker setups. Countless people have better stereo's than many engineers of sound, so long as the signal integrity is there, it's probably going to be good. It might even be to our benefit that they can't as easily obsess over the minutia.

Your project is a big one. But it's an important type of step. You can tweak all day, and pay for it like you've noted, or you can fundamentally upgrade something and get a big pay-out (then tweak it later!).

[Danny Richie]

You bet I will post pictures, I am friends with 2 recording engineers and they always making fun to me, because they say you audiophiles don't want to equalize or DSP but you don't realize that we equalize and DSP the hell out of every recording we work on and most of the best studio monitors are Digital and use DSP, lol I think he have a point. My personal opinion is that yes there is something special about analog! But I got tired of spending thousands equalizing my audio system by changing cables, replacing components or by buying the latest tweaks, is a very expensive and unpredictable way to fine tune your system, you want to fine tune or equalize your system buy an equalizer not 10 different pairs of interconnects lol.

What's done during mixing and what is done during playback is completely different things. They record in in a digital domain and they alter it or EQ it still in the same domain. That is not the issue. The issue is converting it to analog. And most of those recording engineers and the guys doing the mixing never hear what they have recorded. At least they never hear it played back at a level most of the audiophiles out there can recreate it.

Consider that a decent D/A converter will cost somewhere in the $1,000. And most of the great bang for the buck high quality D/A converters are in the $2,000 to $3,000 range. Then there is the elite units that are upwards of $20,000 to $30,000. And that is just for two channels of output. Take that same $2,000 DAC and multiply that times six or eight channels and you are in the $8,000 to $10,000 range real quick and that doesn't even include the digital processing and software used for manipulation. So it is kind of naive to think that you can get the whole package and have great sound for a few hundred dollars.

The DEQX is what I'd consider a good mid-fi level piece if gear and it is in the $4,500 range or more.

Rich Hollis has the best out there that I have heard and the only one that I'd consider a high end unit right now, and I think it is about half of the cost of the DEQX unit. So right now it is the best that I know of and the only piece of gear of its kind that I recommend.

[audiogurujax]

I agree that the DSPMusik hardware is more advance Sadly I tried it and as a software engineer I can tell you that the integration between the unit and the Audio weaver software can get tricky sometimes. I talked to Rich and I am considering buying his unit just to try it out one more time. the main advantage of the DEQX is the simplicity, and there are other very advance options out there, Like you said the most critical part is the DAC and output stage, that is why I am using 2 Auralic Vega dacs with my DEQX and using the digital outputs on the DEQX that way all the DEQX is doing is taking a digital signal doing DSP and producing 3 separate digital signals and the digital to analog conversion is done on the Vega's DAC's Trust me the sound difference is night and day vs using the DEQX analog outputs.

I talked to a guy that is Running a Merging NDAC the 8 channel version and Accurate in a PC and he created a very complex and Powerful DSP Crossover. Yes the DSPMusik is good but its internal DAC's are not a match to having a nice multi-channel DAC like a Lynx or a NDAC or separate DAC's for (low, Mid and High) and Ideally you will have DAC's that allow you to synchronize their clocks so they all run using the same clock. There are Options out there but like the DSPMusik non of them are not straight forward have a Preamp and the functionality and ease of use of the DEQX, but like I said if you used the Digital outputs of the DEQX and add 2 or 3 nice DACs in the mix then I Put the DEQX against any contender. The only problem is that It cost a lot!

Maybe the Option is build it yourself by Buying the DSPBlock from Danville Signal adding a High End Linear PS and have only digital inputs and outputs and use high end DAC's, and if you need analog input add a studio grade ADC. trust me that will be an amazing unit a simple Digital to Digital DSP, and it will be cheap to build!