Mikets42 which older one have you used?
And on what they are weak?
If you don't define then it's just assumption. And here again i can tell from experience it's just plain wrong ( and can explain why).
I was just going by your "if you have an extra $1000"... (wouldn't have replied otherwise).
I just counted about twenty Q-Sys Core110f's that sold for under $1000 since July1. Some for well under $1000.
Q-Sys customer support is truly excellent....open a case and you get a quick reply back. You do not have to be the original owner.
Online Training is even better. https://training.qsc.com/mod/page/view.php?id=560
The schematic design software has about as simple and straightforward drag and drop UI as I've ever seen....
I don't know about other pro-audio brands, but I have to believe they can't be too far away from q-sys, in either direction.
Seriously, all your issues with any reasonably recent open architecture DSPs are mainly ghosts, imho. I know they are for q-sys.
Well if you want something stable and reliable you should not consider multitask ever. That's my observation of digital audio systems i use since mid 90's. Hard to hear for amateurs but it is what we do in pro world. Ymmv but you would had been warned. 😉
Horsepower for convolution? Any computer with 1ghz processor can do convolution even on 192khz filters. The issue which might come is latency but even a 100ghz processor won't change the fact FIR needs delay.
About the 'state of your pc' it's too vague to understand what you are talking about.
I understand the point about amount of cables but it's true whatever you use, pro or not. It can usually be managed with a bit of planning.
Simple graphical interface offered by Hypex or Mini-dsp? Well if those tools are not used in P.A. context it's mainly because they are not simple to operate...
That said User Interface is something you learn by experience with the unit, but in some case i agree some UI doesn't fit people. It's preference related and as such can't be discussed.
And you think this is not possible with a unit which have to manage a FOH+fill+zone(s)+...
If so i doubt you ever used a pro unit.
Then it's a real issue and you should ask about it, we might help you solve this issue using the required technique to do so. Imo, measurements have to be understood if you want to play with a dsp or you are shooting in the dark.
Well you've got no issue if the system had been measured and put in spec using your dsp. Mainly because eq are all over the place regarding Q parameters and the way they apply 'curves' on a signal: there is no standard. It's true for analog but digital too.
It's where measurements rules: once you have your profile acquired, it's just a mater to play with eq parameters until it fits your measurements then it's done.
You are right to struggle with the idea to put a mic 1m in front of your loudspeaker in your room to get useful results. It's only a first step and you need to do more things to have significant results in a domestic room ( gating is needed to keep room interaction at minimum).
But for low end ( your woofer) other technique can be implemented which will take your room out of the equation: close micing. It have limitations but if you know them it can work pretty well. How wide is the cabinet?
Why would you use a signal matrix for something as trivial as selecting an input? In my view you are mixing two principle: hardware management ( selecting inputs) and routing ( what you call matrixing) which define signal path within software/hardware outputs.
Let me take the example of HLC: vst plug in perform filtering that's all.
All routing/matrixing are done by the Host application ( software which 'play' the VST) as well as hardware management.
There is many ways to 'play' a vst, ranging to very simple host to a DAW with almost infinite possibility regarding routing of signal.
Depending on your needs you'll make a choice of Host but it have to be understood the more complex routing are just... complex and might require a DAW which might be complex to set up and use.
That said, VST are now routinely used for live acts being it being a concert, conference, theater,... for such use some host exist like Audioström's Liveprofessor2 which offer powerful routing but relatively manageable in real time despite being powerful with matrixing. I suggest you take a look at what it can offer and think about what it could do used with something like HLC.
Understood and agree.
But you don't have to do riaa in software: i have a physical riaa preamp which is permanently patched to 2 of the 4 inputs of my LMU which act as an AD converter. In fact it ( LMU) act as a soundcard to enter software i use for restauration work. To have access to it i just have to drop a menu and select the input i want in the software. No more hassle than with any other input. And if i want to play direct to loudspeaker i can do the same in the LMU ( select analog pair 1/2 in spite of digital one from computer or CD player or digital desk...
D
Deleted member 375592
I would not want to go on a bragging tour but I used to design them, and ... I have some idea of what they can do.
Wow! Lots of things changed since I retired from pro-audio. It was pretty bad back in the 00s.
There are probably 3 classes of MiniDSP (or any DSP) user.
1. Those that don't measure and simply slap in a "linkwitz this or butterworth that" crossover then tweak to their hearts content until it "sounds good"
2. Those that measure the final in room response and apply an EQ curve. This may end up being better than #1 but you are optimising for one listening position or axis (above the bass localisation frequency)
3. Those that design the crossover for the speaker first (without room effects) based on real measurements. They may then apply some bass equalisation to handle bass room modes.
Camp #3 I think if optimal.
With the speaker design completed, then the decision on amplification simply comes down to how much distortion or non-linearity you like in your amps. Most of the Class AB amplifier reviews I've read show higher distortion than their Class D counterparts. Here's an example:
https://www.audiosciencereview.com/.../yamaha-a-s701-stereo-amplifier-review.55211/
That may explain the "warmer sound" you mentioned and why many may consider some very low distortion Class D amps "cold". They are just accurate.
For me, the optimum is a neutral / linear / low distortion setup. Mostly likely achieved with Class D (based on $/clean watt). Then I can use DSP to add any manner of effects to make it sound "better" and have this switchable on an album by album (or even song by song basis if really needed). This includes EQ and could also include even order distortion, reverb etc... filters if I really want to.
1. Those that don't measure and simply slap in a "linkwitz this or butterworth that" crossover then tweak to their hearts content until it "sounds good"
2. Those that measure the final in room response and apply an EQ curve. This may end up being better than #1 but you are optimising for one listening position or axis (above the bass localisation frequency)
3. Those that design the crossover for the speaker first (without room effects) based on real measurements. They may then apply some bass equalisation to handle bass room modes.
Camp #3 I think if optimal.
With the speaker design completed, then the decision on amplification simply comes down to how much distortion or non-linearity you like in your amps. Most of the Class AB amplifier reviews I've read show higher distortion than their Class D counterparts. Here's an example:
https://www.audiosciencereview.com/.../yamaha-a-s701-stereo-amplifier-review.55211/
That may explain the "warmer sound" you mentioned and why many may consider some very low distortion Class D amps "cold". They are just accurate.
For me, the optimum is a neutral / linear / low distortion setup. Mostly likely achieved with Class D (based on $/clean watt). Then I can use DSP to add any manner of effects to make it sound "better" and have this switchable on an album by album (or even song by song basis if really needed). This includes EQ and could also include even order distortion, reverb etc... filters if I really want to.
Yes they really have.
Back around 2000 i bought a Peavy Media Matrix for a surround sound setup. Pretty dang advanced unit for the time.
Then, in like 2016-18, i got into minidsp opendrcs, and then better spec units like the Linea ASC-48 mentioned (nice unit btw) and some Midas gear.
But i swear, the q-sys stuff has changed my world. It is simply ******** amazing how it allows experimentation...with all the help and resources online to make it easy.
D
Deleted member 375592
Good to hear! The configuration process used to be a pain in the neck and drive you nuts. I would rather write what I want in assembler than **** with UI.
BTW, do you or anybody else have any experience with modules by Nvarcher sold on AliExpress? They use ADSP-21489 on 450MHz (900 MMACS) which is a serious power.
BTW, do you or anybody else have any experience with modules by Nvarcher sold on AliExpress? They use ADSP-21489 on 450MHz (900 MMACS) which is a serious power.
Hi Everyone,
Since the Danville dspNexus is being discussed here, I will jump in. I will try to avoid the sales pitch. You are welcome to set up a video call with me if you like. There is a contact form on the Danville website.
I believe that a competent DSP active crossover can always out perform a passive system. As an example, we did a passive/active A/B comparison using a Magnepan 1.7i (their passive crossover) with our active version at Axpona this year. All components were identical except the crossover section. We didn't have a single person tell us they preferred the passive version. You can find several youtube videos about the demo.
The dspNexus 2/8 is typically 2 inputs and 8 outputs. We can provide it in 4 channel versions..
The latest version uses an Analog Devices ADSP-21569 SHARC which is quite a bit faster than the 4th generation devices used by most everything else that uses a SHARC. We are just in the process of releasing this version. The dspNexus uses AK5578 ADC and AK4499EX DACs.
Audio Weaver is included with the dspNexus It requires a custom platform file that connects the hardware to the software. This is part of our package. We provide technical support.
Thanks for contributing to this conversation. It is very interesting to see how fast acceptance of DSP based digital audio is changing long held assumptions in high performance audio.
Regards
Al Clark
CEO
Danville Signal
Since the Danville dspNexus is being discussed here, I will jump in. I will try to avoid the sales pitch. You are welcome to set up a video call with me if you like. There is a contact form on the Danville website.
I believe that a competent DSP active crossover can always out perform a passive system. As an example, we did a passive/active A/B comparison using a Magnepan 1.7i (their passive crossover) with our active version at Axpona this year. All components were identical except the crossover section. We didn't have a single person tell us they preferred the passive version. You can find several youtube videos about the demo.
The dspNexus 2/8 is typically 2 inputs and 8 outputs. We can provide it in 4 channel versions..
The latest version uses an Analog Devices ADSP-21569 SHARC which is quite a bit faster than the 4th generation devices used by most everything else that uses a SHARC. We are just in the process of releasing this version. The dspNexus uses AK5578 ADC and AK4499EX DACs.
Audio Weaver is included with the dspNexus It requires a custom platform file that connects the hardware to the software. This is part of our package. We provide technical support.
Thanks for contributing to this conversation. It is very interesting to see how fast acceptance of DSP based digital audio is changing long held assumptions in high performance audio.
Regards
Al Clark
CEO
Danville Signal
Did you take any measurements or compare transfer functions? In other words, how did you determine the only difference was passive vs. active rather than XO topology X vs. Y (with different transfer functions and phase characteristics)?
D
Deleted member 375592
"Audio Weaver is included with the dspNexus It requires a custom platform file that connects the hardware to the software. This is part of our package. We provide technical support."
How do you position yourself relative to traditional suppliers of DSP h/w like Qsys and Chinese AliExpress boards?
Do you have resources to provide technical support like Qsys? These have teams of tens of people in support, s/w, f/w, and h/w, in-house manufacturing/QA, and provide a stable, proven, and polished GUI well beyond the awkward SigmaStudio. The cost of components is negligible compared to the price. Chinese boards are a pain in the neck, with zero support (the only support is the threads of this forum), but are dirt cheap, with very low logistical overhead.
Can you provide just boards with popular configurations, like 12dBu 2 in, and 8 out on the price point below ADI's eval boards, for the DIY market via Amazon or PartsExpress? In other words, what is your MSRP-to-BOM ratio?
How do you position yourself relative to traditional suppliers of DSP h/w like Qsys and Chinese AliExpress boards?
Do you have resources to provide technical support like Qsys? These have teams of tens of people in support, s/w, f/w, and h/w, in-house manufacturing/QA, and provide a stable, proven, and polished GUI well beyond the awkward SigmaStudio. The cost of components is negligible compared to the price. Chinese boards are a pain in the neck, with zero support (the only support is the threads of this forum), but are dirt cheap, with very low logistical overhead.
Can you provide just boards with popular configurations, like 12dBu 2 in, and 8 out on the price point below ADI's eval boards, for the DIY market via Amazon or PartsExpress? In other words, what is your MSRP-to-BOM ratio?
Me wonders if the driver "looking back into" the passive crossover impedance has something to do with it. Certainly it doesnt "see" the amplifier output impedance as it would when directly connected; the Xover moved upstream, north of the amplifier.
Way back when there certainly were crossovers realized using active circuits, placed upstream in this way. Why didnt that way of architecting a multi-way system just stomp the speaker level Xover method to death back then? Cost of amplifiers these days?
Audiophiles like to change equipment and have different combination.
With the active analog crossover you mentioned it's not as easy as traditional passive crossover to play with.
Certainly there's still commercial product using this design like the B&W Nautilus but it's not popular.
Studio monitors have more this kind of design.
With the active analog crossover you mentioned it's not as easy as traditional passive crossover to play with.
Certainly there's still commercial product using this design like the B&W Nautilus but it's not popular.
Studio monitors have more this kind of design.
If you are talking a discrete full up analog active system with EQ it's a lot of gear. A stereo amp for each driver once you get beyond bi-amping you really need racks. You are also talking class AB amps so heavy as hell compared to now with heat to manage.
I would say cost and sheer size. With analog you certainly didn't have the flexibility of DSP and had to use well behaved drivers. You had 31band EQ and 1/8 octave from 100Hz or so down and to a limited degree parametric.
I don't buy into the audiophile can't tweak it. You have so damn much flexibility you change amps, crossovers, EQ, drivers, cables it's way more than any passive system. Not getting into the subtle changes in EQ, driver level, and shifting crossover frequencies.
Rob 🙂
You are complety right....it is an awfull lot of gear....but analog XO's + separate power amps + specific chosen drivers are at least for me the way forward.
Of course this is strictly personal. Please no flaming thread "which is the best" because i've witnessed great executions of DSP/class-D audio gear but i ( = personal) simply missed something in those rigs.
My set currently comprises of:
With this set up i can tune, adapt, explore and enjoy. Just took out the 35TG amps (used for the 10" mid/bas ) and swapped it for a First Watt F8 (as the F1 is absolutely remaining to power up the 15" woofers).
Of course i did not start here. I'm a diy tubeamp-builder by nature and had great fun making an over-the-top TB3/1000 amplifier (the European equivalent of the venerable 833). Great result but after finishing that i noted that this great amp was very good in bas/mid but performed lesser at the higher notes. So i build another amp to cover that and did a passive XO to do the formal division. That worked but simply changing an amp (...continious building fun stuff...) or trying to optimise immediately meant to change that passive XO. And proper XO-parts are not for free. At that time i tried active crossovers (DBX and Behringer) which gave me a wealth of options (hurrah !) but left me uninspired by sound-quality (just lost a dimension....). I switched to First Watt B-gear and a Bryston 10B to bring back the analogue part and noted that soundwise this was the preferred way forward for me. But both Bryston and the B-XO's have little options to change which was what i was looking for: to set up the used drivers in their most lineair part (and no funny impendance curves...). The Pass Labs XVR-1 to the rescue ! It took me a couple of years to fetch a stash of those as i needed 3 of them (for a 4-way system).
When using those XVR-1's you work analogue (YES !) but have enough options to tinker around with curves, frequencies, non fixed frequencies for both high pass / low pass . So now i could optimise the drivers work ranges optimum in combination with my DIY- tube power amps.
But.....those XVR-1's work OK single ended but shine when used in balanced mode. Till that time all my gear was single ended (my diy TDA1541 DAC and my diy PT8/101F preamp) but the gain for balanced mode was that easy to hear......switched over to XLR all the way. Enter the XP20 and XP25 preamp and phonostage.
The drivers themselfes changed also in the last years. In order to have maximum flexibility i use fieldcoils throuout. It's a lot of extra power supplies but now you can dail in the characteristic of the driver to the K-tube or open baffle "enclosure"
And this leaves me with an extremely flexible rig: i can "throw" in another amp and dial in the XO's when needed. Just added a First Watt F8 instead of the 35TG amps. Fun....because you can.
Or hear the change between a F1 and F8 ....
So going back to the first sentence: yes...this is a lot of gear. But also a lot of flexibility. And the set on its own is more than capable of producing a proper soundstage/sound-experience.
At this stage i'm really really tempted for a SIT-4 (but First Watt products are impossible to audition in Europe.....) because that F1 seriously impressed me. It gave my 15" woofers a "control" a could not make with diy-tube gear. The F8 gave me another look at First Watt and i see/hear those products work.
And going back to the ChatGTP-opening: i do like that classical old school class A amplifier + analogue amps etc etc.....It's not the easiest way forward but give nice sounding results.
Regards, Reinout
Of course this is strictly personal. Please no flaming thread "which is the best" because i've witnessed great executions of DSP/class-D audio gear but i ( = personal) simply missed something in those rigs.
My set currently comprises of:
- Pass Labs XP20 preamp
- Pass Labs DAC1
- Bel Canto CD2 cd-transport
- Horstmann & Petter Ulysses turntable + Rigid Float tonearm + Denon 103 extreme modified cartridge
- Pass Labs XP25 phono-stage
- 3 x Pass Labs XVR-1 XO
- diy Eimac 35TG power amp (8 watt...)
- diy 6C45 power amp (2.6 watt...but 2600 milliwatt sounds more impressive....)
- diy Ce/ED111 power amp (700 milliwatt...no kidding)
- First Watt F1 transconductance power amp
- diy 4 way full fieldcoil loudspeaker system (mid and high are compression drivers with K-tubes): 15" + 10" + 2" + 1")
With this set up i can tune, adapt, explore and enjoy. Just took out the 35TG amps (used for the 10" mid/bas ) and swapped it for a First Watt F8 (as the F1 is absolutely remaining to power up the 15" woofers).
Of course i did not start here. I'm a diy tubeamp-builder by nature and had great fun making an over-the-top TB3/1000 amplifier (the European equivalent of the venerable 833). Great result but after finishing that i noted that this great amp was very good in bas/mid but performed lesser at the higher notes. So i build another amp to cover that and did a passive XO to do the formal division. That worked but simply changing an amp (...continious building fun stuff...) or trying to optimise immediately meant to change that passive XO. And proper XO-parts are not for free. At that time i tried active crossovers (DBX and Behringer) which gave me a wealth of options (hurrah !) but left me uninspired by sound-quality (just lost a dimension....). I switched to First Watt B-gear and a Bryston 10B to bring back the analogue part and noted that soundwise this was the preferred way forward for me. But both Bryston and the B-XO's have little options to change which was what i was looking for: to set up the used drivers in their most lineair part (and no funny impendance curves...). The Pass Labs XVR-1 to the rescue ! It took me a couple of years to fetch a stash of those as i needed 3 of them (for a 4-way system).
When using those XVR-1's you work analogue (YES !) but have enough options to tinker around with curves, frequencies, non fixed frequencies for both high pass / low pass . So now i could optimise the drivers work ranges optimum in combination with my DIY- tube power amps.
But.....those XVR-1's work OK single ended but shine when used in balanced mode. Till that time all my gear was single ended (my diy TDA1541 DAC and my diy PT8/101F preamp) but the gain for balanced mode was that easy to hear......switched over to XLR all the way. Enter the XP20 and XP25 preamp and phonostage.
The drivers themselfes changed also in the last years. In order to have maximum flexibility i use fieldcoils throuout. It's a lot of extra power supplies but now you can dail in the characteristic of the driver to the K-tube or open baffle "enclosure"
And this leaves me with an extremely flexible rig: i can "throw" in another amp and dial in the XO's when needed. Just added a First Watt F8 instead of the 35TG amps. Fun....because you can.
Or hear the change between a F1 and F8 ....
So going back to the first sentence: yes...this is a lot of gear. But also a lot of flexibility. And the set on its own is more than capable of producing a proper soundstage/sound-experience.
At this stage i'm really really tempted for a SIT-4 (but First Watt products are impossible to audition in Europe.....) because that F1 seriously impressed me. It gave my 15" woofers a "control" a could not make with diy-tube gear. The F8 gave me another look at First Watt and i see/hear those products work.
And going back to the ChatGTP-opening: i do like that classical old school class A amplifier + analogue amps etc etc.....It's not the easiest way forward but give nice sounding results.
Regards, Reinout
Hi, If I lived in the US I would be very interested in the dspNexus. But import taxes in Denmark are just insane 😫
Getting the Magnepan to sound better is not that hard. The make really bad filters. Too much overlap in the frequency's bands.
Guess you also used a sharper filter?
Don’t take this the wrong way but you guys really make a full range speaker look good. Even with setting up digital room correction someone with a single driver system and a stereo set-up is hours into listening to the music before most of you DSP geeks have everything unboxed, much less plugged in. Seriously, for surround sound do you have 23 or more amplifiers? Ya’ll are pretty lucky, my wife can’t stand the cables for five plus one speakers, and your cats must have a great time playing behind your audio system. I think a stereo system with DSP, four amplifiers, and two way speakers is the way to go, anything else and you are obsessed with the technology and not the music. I guess that’s OK, too. Like a guy with a big 4X4 who never gets it muddy. One of the best systems I ever heard was a five channel set up with elecrostatics and JBL subwoofers in concrete boxes. It was a two way system with electronic XO’s and ten bigassed amps in an isolated room. Very impressive. Something Bruce Wayne would have.
A thought here, is it better to match amplifiers or purposely un-match them? Would the system sound better with tubes up top, class A/B for the mids, and class D for the bass?
A thought here, is it better to match amplifiers or purposely un-match them? Would the system sound better with tubes up top, class A/B for the mids, and class D for the bass?
To my mind this is just one of those tech shifts that are inevitable but acceptance is slow. In Europe and the UK people were resisting multi-channel tv for decades, once the ratchet moved there was no looking back. Same for mobile phones in the US and then smartphones - etc, etc.
If you resist plate amps you can readily get small multi channel amps equipped , with f.i. Pascal modules that will outperform most “audiophile” amps, and place them next to your speakers with a balanced analog cord to each speaker and end up with a cleaner setup than w a passive system.
If you resist plate amps you can readily get small multi channel amps equipped , with f.i. Pascal modules that will outperform most “audiophile” amps, and place them next to your speakers with a balanced analog cord to each speaker and end up with a cleaner setup than w a passive system.
Ok but after looking at the video didn't you say you changed the speakers to measure flat on axis? You changed the crossover. If you altered the original voltage drive to the different panels that is obviously going to voice them differently. If that's the case it's not a 1 to 1 comparison.
Did I understand that right?
Rob 🙂